- sr-users - kamailio.org

[Users] record routing in dispatcher
by Rick Richardson 06 Mar '06

06 Mar '06

Hello group, I am trying to determine if record routing would ever be useful on a dispatcher type sip router. It would seem to me that as long as the dispatcher hashes on the call-id then it will always ensure that the proper proxy receives the message. I do notice in rfc 3261 that it says, "The proxy will remain on the path if it chooses not to insert a Record-Route header field value into requests that are already part of a dialog. However, it would be removed from the path when an endpoint that has failed reconstitutes the dialog." In the case of reconstituting a dialog, does it keep the same call-id? I can't find anything in the rfc that says it does. This would imply, then, that the dispatcher should handle record routing for cases such as this. So that it can make sure to forward to the proper proxy even when the dialog temporarily breaks down. On that subject. I notice that most of the RR examples I see use: if (loose_route()) { t_relay(); } or something similar. Is there a way to forward the message on to the proper proxy using rr without using the tm module? Or perhaps a better question would be: what is the proper way to forward on to the correct box with record routing? thanks, Rick

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[Serusers] How to make SER keep states
by M＠rcio 06 Mar '06

06 Mar '06

Hi, I'm working with SER and I'm trying to make a state machine for SER. I'm having some difficults on trying to understand how modules work, because SER have to be extended to enable such functionality. The idea is to SER recognize that a client has already estabilished a conection with a video server throught SER's proxy and therefore SER must redirect the new INVITE to another proxy. In this new proxy the trafic will be multicast, so that everyone who wants the same video service will have it. SER just need to recognize that two or more clients wants to estabilish conection with the same terminal. Can you help me in how or where in the modular arquitecture of SER will I have to act? Thanks.

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[Serusers] cdrs?
by roger leszczynski 06 Mar '06

06 Mar '06

does anyone have example files to track cdrs in flat file and mysql database? im also wondering what happens iof i restart ser while 100 concurrent calls are occuring, and they are still going through rtpproxy, will the acc module keep tracking those cdrs?

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[Users] Custom header field use || getting original IP from dispatcher
by Rick Richardson 06 Mar '06

06 Mar '06

I am needing to use the source IP for a few routines in my proxy server, but the problem is that it is being forwarded to me by a dispatcher. I am definitely needing the IP address. My first inclination was to append a header field with the IP address. This is quite easy. However, the hard part is using that passed in IP address. I'm not sure what the best way would be to get that value into the hands of several module functions like permissions => allow_trusted(), etc.. Any ideas? thanks, Rick

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[Serusers] Fw: [Sems] Sems Thread Memory
by Nicky 06 Mar '06

06 Mar '06

Hi Don, For what I see from the system monitoring screen, the sems thread was removed when a caller hang up but the main sems threads are still running and growing in size. When the size reach 2GB, I can see pthread_create error 11. Size will only shrink when I restart sems but all my calls would have to be terminated abruptly. I'm not familiar with ethereal in linux, is there an easier way to solve this, like sample codes that I can use in the script? I have increased the RAM size to 1GB but it didn't help much. Sems just couldn't take more than 10 calls together running for more than 2hrs. Any advice is greatly appreciated. Thanks. Regards, Nicky ----- Original Message ----- From: Nicky To: don.fletcher(a)att.net ; serusers(a)lists.iptel.org ; sems(a)lists.iptel.org Sent: Wednesday, February 22, 2006 9:37 AM Subject: Re: [Sems] Sems Thread Memory Hi Don, Thank you for your reply. How can I check using commands that the thread is still in session? What are the commands in the scripts that close the rtp session? Please advice. Thanks Regards, Nicky ----- Original Message ----- From: don.fletcher(a)att.net To: Nicky Sent: Tuesday, February 21, 2006 2:47 AM Subject: Re: [Sems] Sems Thread Memory Nicky, Probably the RTP streams are not being closed down properly at the end of a call so that the threads remain in existence. Look at the media streams with ethereal. Look closely at call disconnect. Be sure all threads are closed. don -------------- Original message from "Nicky" <nicky(a)caliber.com.sg>: -------------- Dear All, I am having a serious problem, my thread memory size is growing to up to 2GB which applies to SEMS, SER and Mysqld thread. Please help. I am testing with about 10 people calling in at the same time. What shall I do? :( Nicky ----- Original Message ----- From: Nicky To: serusers(a)lists.iptel.org ; sems(a)lists.iptel.org Sent: Saturday, October 29, 2005 11:44 AM Subject: pthread create failed with code 11 Hi Support, I have experienced this error, please advice what can be the caused. Thanks. ERROR: start (AmThread.cpp:85): pthread create failed with code 11 Regards, Nicky

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[Users] Does CDRTool support SIP REFER?
by A B 06 Mar '06

06 Mar '06

Hi All, I'm curious if anyone knows if CDRTool can properly handle call transfers made with SIP REFER and bill the referrer accordingly? On a related question, can CDRTool handle "302 Moved Temporarily" call forwarding? Any info would be greatly appreciated. Thanks! __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Modify SER via serctl or DB?
by Nick Hoffman 05 Mar '06

05 Mar '06

Hi guys, I'm going to be writing a script to make modifications to SER (Eg: add/modify/delete users/aliases, etc). I'm wondering whether it's "better" to 1) use /usr/sbin/serctl or 2) directly modify the MySQL SER database What advantages/disadvantages are there for each method? Are there any reasons why it's not advisable to directly modify the MySQL SER database to accomplish tasks such as adding a user? Thanks for your input! -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

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[Serusers] not registering
by alex 05 Mar '06

05 Mar '06

i have set ser up on suse and it all seems to be working fine itsthe newest version of ser with mysql all tables went in fine i edited the config file and there were no errors i was abel to add users useing the serctl command e.g. serctl add username pass email and it went in fine i checked it with phpmyadmin but when i try and ceonect to my server with the corect username and pass it conects but it dosent register here is the log from express talk 15:27:43 Restarting the SIP Engine so settings take effect. 15:27:50 SIP Private IP is: 192.168.1.3:5060 15:27:50 Unable to auto-detect public IP. Using private IP 15:27:50 SIP Public IP is: 192.168.1.3:5060 15:27:50 SIP Number: admin@192.168.1.3:5060 15:28:30 Server 192.168.1.5 did not respond to register (user admin) 15:28:30 Check server details for that line useing ethereal on the server it shows the request comeing in but the server dosent respond. i dont know if ser keeps any logs andf if they do i have no idea were they are. im sure this is a comman error so it should hopefully be easy to fix. :) thanks

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[Serusers] updated FreeBSD port
by Hendrik Scholz 05 Mar '06

05 Mar '06

Hi! I managed to update the FreeBSD SER port during some longish friday afternoon phone conference. - rm -r /usr/ports/net/ser - extract http://dump.wormulon.net/patches/ser-freebsd.tar.gz to /usr/ports/net (will create ser directory) I've added a bunch of OPTIONS. If you need any more drop me a mail. The init script part is still untouched and as of now you need to copy the init script to /usr/local/etc/rc.d et al. The port is based on the CVS freebsd/ directory rather than the FreeBSD port itself wich is cluttered with small patches and appeared hard to update/maintain. Enjoy, Hendrik -- Hendrik Scholz - <hscholz(a)raisdorf.net> - http://www.wormulon.net/

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[Serusers] TURN and open source implementation
by Vikas Sarawat 05 Mar '06

05 Mar '06

Hello, Does IPTEL.org have an open source implementations available for TURN? Please let me know if you guys know of any. Thanks in advance, Vikas

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[Users] avpops: new function avp_db_query()
by Daniel-Constantin Mierla 04 Mar '06

04 Mar '06

Hello, avpops module has a new function which allow to execute raw SQL queries and store the result in AVPs. avp_db_query(query, dest); The query given as parameter can contain pseudo-variables. Using this function you can benefit of full database system features, being able to do joins, unions, etc. Old db-related functions are in place since they are faster for their usage case. The documentation of the of avpops module was updated and posted at: http://openser.org/docs/modules/1.1.x/avpops.html A small example of usage: limit the number of calls done in the last day: if(is_method("INVITE") && !has_totag()) { if(avp_db_query("select count(*) from acc where username='$fU' and domain='$fd' and method='INVITE' and timestamp>=$Ts-24*3600", "$avp(i:234)")) { if(avp_chech("$avp(i:234)", "ge/i:10")) { sl_send_reply("403", "too many calls in the last day"); exit(); } } } Cheers, Daniel

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Re: [Users] how to route a call between servers
by Daniel-Constantin Mierla 04 Mar '06

04 Mar '06

On 03/03/06 07:46, vikram sarathy wrote: > > hi daniel, > thanks for this mail, i created a table in database and stored > ip's of all servers and by using avp_db_query() function, and trying > to store all this ip's of server into default varables > "$avp(i:1);$avp(i:2);...". > in my openser.conf file i am trying to do, > rewritehost("$avp(i:1)"); > rewritehost() has no support for pseudo-variables. You have to use avp_pushto("$ruri/domain","$avp(i:1)"). Cheers, Daniel > append_branch(); > rewritehost("$avp(i:2)"); > append_branch(); > for parrallel forking. here i have used 2 ip's hard coded. this has to > happen in generic, mean to say for the number of servers in database > list,rewritehost() and append_branch() has to called. is there any > function to do this. > > with regrads > vicky > > > On Thu, 02 Mar 2006 Daniel-Constantin Mierla wrote : > >Hello, > > > >On 03/02/06 10:23, vikram sarathy wrote: > >> > >> hi, > >> i have 3 servers in my setup, in which each server has a user > registered. i need to route a call from one server to other. in first > case i kept all servers in same domine and later i need to check with > all servers in different domains. > >> > >>can any one tell me is there any way to achive this. > >> > >one possible way, very simple, is to store for each user the address > of the server which keeps the location information. You can load the > address via avpops module and then set the dst_uri to it and forward. > > > >Other way is to split the users based on some rules (e.g, prefix) to > learn easily to which server the user belongs. As an alternative, you > can try serial/parallel forking to all servers. > > > >Daniel > > > >> > >>with regards > >>vicky > >> > >>vicky > >> > >> > >><http://adworks.rediff.com/cgi-bin/AdWorks/sigclick.cgi/www.rediff.com/signa…> > > >>------------------------------------------------------------------------ > >> > >>_______________________________________________ > >>Users mailing list > >>Users(a)openser.org > >>http://openser.org/cgi-bin/mailman/listinfo/users > >> > > > > <http://adworks.rediff.com/cgi-bin/AdWorks/sigclick.cgi/www.rediff.com/signa…>

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[Users] LCR module and call forwarding
by Florian Fainelli 04 Mar '06

04 Mar '06

Hello, I am attempting to use the lcr module (well configured, routes are found), but I can't get it forward my calls to the destination gateway. - is it implicit that load_gws() in a route block will forward calls ? - if not, how do I manage to get calls forwarded ? Thank you very much for your answer. -- Cordialement, Florian Fainelli --------------------------------------------- 5, rue Charles Fourier Chambre 1511 91011 Evry http://www.alphacore.net (+33) 01 60 76 64 86 (+33) 06 09 02 64 95 --------------------------------------------- Institut National des Télécommunications http://www.int-evry.fr/telecomint ---------------------------------------------

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[Serusers] Two clients behind same NAT and external registrar
by Chris Roberts 04 Mar '06

04 Mar '06

Heya, I've found this code for detecting if two clients are behind the same NAT from the AVPops documentation: ... # get to the final RURI if (!lookup("location")) { sl_send_reply("404","Not found"); break; } # get the host part of the final uri (IP part) and store it in AVP ID 13 avp_write("$ruri/domain", "i:13"); if (avp_check("i:13","eq/$src_ip/i")) { # source IP is the same as destination IP ........... } avp_delete("i:13/g"); ... I'm trying to apply this in my situation with an external registrar, and I can't think of a way I can make this fit. The basic call flow is: UAC->SER (INVITE) SER->REGISTRAR (INVITE) REGISTRAR->SER (Trying/302 Moved) SER->UAC (INVITE) UAC->SER (Trying/OK) SER->UAC (ACK) This uses the Contact header in the returned response from the registrar to provide serial forking, etc. I can't see how to make this work - the only thought I have is that I may be able to leave the original SDP header intact on the invite to the registrar, and then only fix the SDP header in the INVITE sent to the UAC AFTER the 302 moved, by using the onbranch_route (as we should now have both dest/source). I think I tried something like this though and it didn't work. Any ideas, is my logic correct? Cheers, Chris. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.1.2/274 - Release Date: 03/03/2006

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[Serusers] ser with radius group checking - something amiss
by Velimir Novkovic 04 Mar '06

04 Mar '06

Hi, I run SER with Radius/MySQL for authentication and accounting. Things are pretty much in place except for group checking. I have something like this in my ser.cfg: .... modparam("auth_radius", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("group_radius", "use_domain", 1) ..... if (uri=~"^sip:[0-9]{8}@") { # Domestic PSTN if (!radius_is_user_in("credentials", "ld")) { sl_send_reply("403", "No permission for domestic calls"); break; }; route(4); break; }; .... When I look at Radius debug log I can see that when ser sends a request to radius, radius wants to do digest on it and then the complete request fails and call can't go through. Output looks something like this: .. rad_recv: Access-Request packet from host 127.0.0.1:34027, id=18, length=72 User-Name = "81000(a)sage.home.local" Sip-Group = "voicemail" Service-Type = Group-Check NAS-Port = 0 NAS-IP-Address = 127.0.0.1 Processing the authorize section of radiusd.conf .. ** bunch of sql statements ... .. modcall: group authorize returns ok for request 17 rad_check_password: Found Auth-Type Digest auth: type "digest" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 17 ERROR: No Digest-Nonce: Cannot perform Digest authentication modcall[authenticate]: module "digest" returns invalid for request 17 modcall: group authenticate returns invalid for request 17 auth: Failed to validate the user. In databases I have following: mysql> SELECT id,UserName,Attribute,Value,op FROM radreply WHERE Username = '81000(a)sage.home.local' ORDER BY id; +----+-----------------------+--------------+-------------+----+ | id | UserName | Attribute | Value | op | +----+-----------------------+--------------+-------------+----+ | 18 | 81000(a)sage.home.local | Service-Type | Group-Check | := | +----+-----------------------+--------------+-------------+----+ 1 row in set (0.00 sec) mysql> SELECT id,UserName,Attribute,Value,op FROM radcheck WHERE Username = '81000(a)sage.home.local' ORDER BY id; +----+-----------------------+---------------+------------------------------ ------+----+ | id | UserName | Attribute | Value | op | +----+-----------------------+---------------+------------------------------ ------+----+ | 23 | 81000(a)sage.home.local | User-Password | $1$d7XAeahG$9f17cb8JaKj8R1z9GpwG4/ | := | | 25 | 81000(a)sage.home.local | Sip-Rpid | 81000 | = | | 30 | 81000(a)sage.home.local | Auth-Type | Digest | := | +----+-----------------------+---------------+------------------------------ ------+----+ mysql> SELECT radgroupcheck.id,radgroupcheck.GroupName,radgroupcheck.Attribute,radgroupche ck.Value,radgroupcheck.op FR OM radgroupcheck,usergroup WHERE usergroup.Username = '81000(a)sage.home.local' AND usergroup.GroupName = radgroupcheck.G roupName ORDER BY radgroupcheck.id; +----+-----------+-----------+--------+----+ | id | GroupName | Attribute | Value | op | +----+-----------+-----------+--------+----+ | 12 | voicemail | Auth-Type | Accept | := | +----+-----------+-----------+--------+----+ Has anyone had a chance to do something like this with success? I am stuck at the moment - any help is greatly appreciated. Thanks. /Vel

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Re: [Serusers] how to modify mysql to get rid of problem running is_from_local()?
by lin9k＠yahoo.com 04 Mar '06

04 Mar '06

Hi George, Thanks for your reply. May I know how you modified the ser database? Ling --- George Lambson <LambsonGE(a)mtc.byu.edu> wrote: > I think there is a way to use serctl to do it, but I > have had success just directly modifying the domain > table of the SER database. > > George > > >>> <lin9k(a)yahoo.com> 3/3/2006 7:11 PM >>> > Hi, > > I'm very new to mysql although have played ser for a > while. Whenever ser.cfg call is_from_local() in > domain > module it would always fail with error log "is > local(): Realm '192.168.100.100' is not local". > > Searching from the emails people suggested to match > the domain name of the client with mysql. But, how > do > I modify mysql to make is_from_local() work? From > serctl? or Directly from mysql>? How? > > Thanks in advance, > Ling > > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam > protection around > http://mail.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [Serusers] Noisy feedback tells
by George Lambson 04 Mar '06

04 Mar '06

SER sends a 200 ok WITH a digest authentication request. The client should then send another REGISTER message with the key that it recieved in the 200 ok. SER should then get the 2nd message and send another 200 ok verifying that the client did use digect authentication. George >>> Piergiorgio Venuti <piergiorgio(a)mediaservice.net> 3/2/2006 3:08 AM >>> Hi all, I've this problem when I start my soft phone client with ser: Warning: 392 192.168.87.252:5060 "Noisy feedback tells: pid=424 req_src_ip=192.168.87.71 req_src_port=6756 in_uri=sip:ser.mydomain.pri out_uri=sip:ser.mydomain.pri via_cnt==1" I look that when my client try to register to ser, it received an 200 ok response, but my client resend a register request again and again, this loop cause a latence in my traffic. Any idea? In attach is my config file Tnx -- +----------------------------------------------------------------------+ | Ing. Piergiorgio Venuti, CCSP Email: piergiorgio(a)mediaservice.net | | System Manager Tel: +39-011-32.72.100 | | Fax: +39-011-32.46.497 | | @ Mediaservice.net S.R.L. http://mediaservice.net | | Via S.Bernardino 17 Torino | | Disclaimer: http://@Mediaservice.net/disclaimer | +----------------------------------------------------------------------+

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Re: [Serusers] how to modify mysql to get rid of problem running is_from_local()?
by George Lambson 04 Mar '06

04 Mar '06

I think there is a way to use serctl to do it, but I have had success just directly modifying the domain table of the SER database. George >>> <lin9k(a)yahoo.com> 3/3/2006 7:11 PM >>> Hi, I'm very new to mysql although have played ser for a while. Whenever ser.cfg call is_from_local() in domain module it would always fail with error log "is local(): Realm '192.168.100.100' is not local". Searching from the emails people suggested to match the domain name of the client with mysql. But, how do I modify mysql to make is_from_local() work? From serctl? or Directly from mysql>? How? Thanks in advance, Ling __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] how to modify mysql to get rid of problem running is_from_local()?
by lin9k＠yahoo.com 04 Mar '06

04 Mar '06

Hi, I'm very new to mysql although have played ser for a while. Whenever ser.cfg call is_from_local() in domain module it would always fail with error log "is local(): Realm '192.168.100.100' is not local". Searching from the emails people suggested to match the domain name of the client with mysql. But, how do I modify mysql to make is_from_local() work? From serctl? or Directly from mysql>? How? Thanks in advance, Ling __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [Serusers] where can i find documents for ser.cfg.
by George Lambson 04 Mar '06

04 Mar '06

Here is the best and most complete information I have found, but with the module documentation I have found a few errors. These are for openser, mostly the same. http://openser.org/docs/modules/devel/ http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook George >>> "wu yaqian" <wflyfox2003(a)gmail.com> 3/2/2006 8:28 PM >>> Hello,erery one I want to write ser.cfg myself.where can i find some documents about this subject. thanks a lot

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FW: [Serusers] Modify SER via serctl or DB?
by Richard C. Thompson 03 Mar '06

03 Mar '06

I have found that in some cases the serctl function doesn't function in mode 2 but does in mode 1, so I have had to write scripts that modify the database directly. As to the advantages and disadvantages, I don't know. My ser is running quite nicely now that I am not writing a lot of comments to the logs. In production, try to keep ser as quiet as possible. RickT -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Nick Hoffman Sent: Friday, March 03, 2006 2:24 AM To: seruser Mailing List Subject: [Serusers] Modify SER via serctl or DB? Hi guys, I'm going to be writing a script to make modifications to SER (Eg: add/modify/delete users/aliases, etc). I'm wondering whether it's "better" to 1) use /usr/sbin/serctl or 2) directly modify the MySQL SER database What advantages/disadvantages are there for each method? Are there any reasons why it's not advisable to directly modify the MySQL SER database to accomplish tasks such as adding a user? Thanks for your input! -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Users] Mediaproxy visible IPs
by Chris Roberts 03 Mar '06

03 Mar '06

Heya, I'm trying to understand the logic in the python side of the media proxy. Specifically, I have a user with a public IP address that is behind NAT. This would seem to be okay, except for this check: elif not caller.signedInAddress and caller.local and address[0]==caller.visibleIP: self.signIn(caller, address, data) sender = 0 destination = called.address elif not called.signedInAddress and called.local and address[0]==called.visibleIP: self.signIn(called, address, data) sender = 1 destination = caller.address I don't really understand the logic of this. Reading some more source, local is defined if the SDP header has a local media stream setting in it's m= header, but I do not understand why this would affect this part of the peer logic. If I removed this local check, my mediaproxy would be able to match incoming RTP streams from users with public IP addresses (visibleIP == public IP) behind NAT to their RTPPeer correctly. But I need to understand the implications of doing this, or whether there might be a better way for me to go? Any suggestions? Cheers, Chris. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.1.1/273 - Release Date: 02/03/2006

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[Users] NEW Functionalities in UAC module
by Bogdan-Andrei Iancu 03 Mar '06

03 Mar '06

Hi, the UAC module from the devel branch includes new functionalities: - you may specify a password for encrypting the original FROM URI as saved as RR parameter. See the new module parameter "from_passwd" http://openser.org/docs/modules/1.1.x/uac.html#AEN9 - uac_replace_from() adds the display name if not present (so far it only replaced it).So, now you can set a display name if none was present. regards, Bogdan

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Re: [Serusers] permission module
by Miklos Tirpak 03 Mar '06

03 Mar '06

Hi Hakan, There was a bug in the postgres module in the last stable branch. I corrected it, so please update your source and recompile postgres module... Miklos Hakan YASTI wrote: > Hi Miklos, > > > I have added as you said. > > select * from trusted; > src_ip | proto | from_pattern > -------------+-------+---------------------- > 84.51.32.26 | udp | ^sip:902247654321@.* > (1 row) > > But it seems same. Could we see the log where the module is comparing > the parameters which are coming with sip message and the result of the > query ? > > ################### ser log #################### > > sertest# 3(71282) SIP Request: > 3(71282) method: <INVITE> > 3(71282) uri: <sip:02124440111@gi.com.tr;user=phone> > 3(71282) version: <SIP/2.0> > 3(71282) parse_headers: flags=1 > 3(71282) Found param type 232, <branch> = <z9hG4bKacAMoJdyS>; state=16 > 3(71282) end of header reached, state=5 > 3(71282) parse_headers: Via found, flags=1 > 3(71282) parse_headers: this is the first via > 3(71282) After parse_msg... > 3(71282) preparing to run routing scripts... > 3(71282) parse_headers: flags=128 > 3(71282) DEBUG:maxfwd:is_maxfwd_present: value = 70 > 3(71282) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 > 3(71282) parse_headers: flags=8 > 3(71282) DEBUG: add_param: tag=1c2131911325 > 3(71282) end of header reached, state=29 > 3(71282) grep_sock_info - checking if host==us: 9==10 && [gi.com.tr] == > [84.51.32.8] > 3(71282) grep_sock_info - checking if port 5060 matches port 5060 > 3(71282) grep_sock_info - checking if host==us: 9==9 && [gi.com.tr] == > [127.0.0.1] > 3(71282) grep_sock_info - checking if port 5060 matches port 5060 > 3(71282) grep_sock_info - checking if host==us: 9==10 && [gi.com.tr] == > [84.51.32.8] > 3(71282) grep_sock_info - checking if port 5060 matches port 5060 > 3(71282) grep_sock_info - checking if host==us: 9==9 && [gi.com.tr] == > [127.0.0.1] > 3(71282) grep_sock_info - checking if port 5060 matches port 5060 > 3(71282) val2str(): converting 84.51.32.26, 11 > 3(71282) PG[217] str2valp got string udp > 3(71282) PG[217] str2valp got string ^sip:902247654321@.* > 3(71282) NOT TRUSTED IP 3(71282) parse_headers: flags=4 > 3(71282) end of header reached, state=9 > 3(71282) DEBUG: get_hdr_field: <To> [40]; > uri=[sip:02124440111@gi.com.tr;user=phone] > 3(71282) DEBUG: to body [<sip:02124440111@gi.com.tr;user=phone> > ] > 3(71282) parse_headers: flags=-1 > 3(71282) get_hdr_field: cseq <CSeq>: <1> <INVITE> > 3(71282) DEBUG: get_hdr_body : content_length=244 > 3(71282) found end of header > 3(71282) check_via_address(84.51.32.26, 84.51.32.26, 1) > 3(71282) DEBUG:destroy_avp_list: destroying list 0x0 > 3(71282) receive_msg: cleaning up > 2(71281) SIP Request: > 2(71281) method: <ACK> > 2(71281) uri: <sip:02124440111@gi.com.tr;user=phone> > 2(71281) version: <SIP/2.0> > 2(71281) parse_headers: flags=1 > 2(71281) Found param type 232, <branch> = <z9hG4bKacAMoJdyS>; state=16 > 2(71281) end of header reached, state=5 > 2(71281) parse_headers: Via found, flags=1 > 2(71281) parse_headers: this is the first via > 2(71281) After parse_msg... > 2(71281) preparing to run routing scripts... > 2(71281) parse_headers: flags=4 > 2(71281) DEBUG: add_param: tag=26a82380ee921ee699cdfa26683b3165.871c > 2(71281) end of header reached, state=29 > 2(71281) DEBUG: get_hdr_field: <To> [82]; > uri=[sip:02124440111@gi.com.tr;user=phone] > 2(71281) DEBUG: to body [<sip:02124440111@gi.com.tr;user=phone>] > 2(71281) DEBUG: sl_filter_ACK : local ACK found -> dropping it! > 2(71281) DEBUG:destroy_avp_list: destroying list 0x0 > 2(71281) receive_msg: cleaning up > > > > ################# ngrep log #################### > > U 84.51.32.26:5060 -> 84.51.32.8:5060 > INVITE sip:02124440111@gi.com.tr;user=phone SIP/2.0..Via: SIP/2.0/UDP > 84.51.32.26;branch=z9hG4bKacAMoJdyS..Max-Forwards: 70..From: > "902247654321" <sip:9 > 02327654321(a)gi.com.tr>;tag=1c2131911325..To: > <sip:02124440111@gi.com.tr;user=phone>..Call-ID: > 1845810222fYtV@84.51.32.26..CSeq: 1 INVITE..Contact: <sip: > 902327654321(a)84.51.32.26>..Supported: > em,100rel,timer,replaces,path..Allow: > REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPD > > ATE..Session-Expires: 3600..Min-SE: 90..User-Agent: > Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.200.371..Content-Type: > application/sdp..Content-Length: 244 > ....v=0..o=AudiocodesGW 585174 818405 IN IP4 > 84.51.32.26..s=Phone-Call..c=IN IP4 84.51.32.26..t=0 0..m=audio 4000 > RTP/AVP 0 8 96..a=rtpmap:0 pcmu/8000.. > a=rtpmap:8 pcma/8000..a=rtpmap:96 telephone-event/8000..a=fmtp:96 > 0-15..a=ptime:20..a=sendrecv.. > # > U 84.51.32.8:5060 -> 84.51.32.26:5060 > SIP/2.0 403 ONLY REGISTERED USERS......Via: SIP/2.0/UDP > 84.51.32.26;branch=z9hG4bKacAMoJdyS..From: "902247654321" > <sip:902327654321@gi.com.tr>;tag=1c213 > 1911325..To: > <sip:02124440111@gi.com.tr;user=phone>;tag=26a82380ee921ee699cdfa26683b3165.871c..Call-ID: > 1845810222fYtV@84.51.32.26..CSeq: 1 INVITE..Serv > er: Sip EXpress router (0.9.3 (i386/freebsd))..Content-Length: > 0..Warning: 392 84.51.32.8:5060 "Noisy feedback tells: pid=71282 > req_src_ip=84.51.32.26 > req_src_port=5060 in_uri=sip:02124440111@gi.com.tr;user=phone > out_uri=sip:02124440111@gi.com.tr;user=phone via_cnt==1".... > # > U 84.51.32.26:5060 -> 84.51.32.8:5060 > ACK sip:02124440111@gi.com.tr;user=phone SIP/2.0..Via: SIP/2.0/UDP > 84.51.32.26;branch=z9hG4bKacAMoJdyS..Max-Forwards: 70..From: > "902247654321" <sip:9023 > 27654321(a)gi.com.tr>;tag=1c2131911325..To: > <sip:02124440111@gi.com.tr;user=phone>;tag=26a82380ee921ee699cdfa26683b3165.871c..Call-ID: > 1845810222fYtV(a)84.5 > 1.32.26..CSeq: 1 ACK..Contact: > <sip:902327654321@84.51.32.26>..Supported: > em,timer,replaces,path..Allow: > REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,P > RACK,REFER,INFO,SUBSCRIBE,UPDATE..User-Agent: > Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.200.371..Content-Length: 0.... > > > > > > > > > > ----- Original Message ----- From: "Miklos Tirpak" <miklos(a)iptel.org> > To: "Hakan YASTI" <hakanyasti(a)gmail.com> > Cc: <serusers(a)lists.iptel.org> > Sent: Thursday, March 02, 2006 3:54 PM > Subject: Re: [Serusers] permission module > > >> Hi Hakan, >> >> Try this value for from_patter without quotes: ^sip:902247654321@.* >> udp is correct for proto >> >> Miklos >> >> Hakan YASTI wrote: >> >>> Hi Miklos, >>> >>> It was output of ngrep -W byline, The original ngrep output is as below. >>> >>>> From fielad is like From: "902247654321" >>> >>> >>> <sip:902247654321@gi.com.tr>;tag=1c572128494. >>> >>> What I wonder is ; according to this invite what must be the proto >>> and from_pattern values to allow the call. >>> >>> >>> U 84.51.32.26:5060 -> 84.51.32.8:5060 >>> INVITE sip:02124440111@gi.com.tr;user=phone SIP/2.0..Via: >>> SIP/2.0/UDP 84.51.32.26;branch=z9hG4bKacGtOiLfA..Max-Forwards: >>> 70..From: "902247654321" <sip:9 >>> 02247654321(a)gi.com.tr>;tag=1c572128494..To: >>> <sip:02124440111@gi.com.tr;user=phone>..Call-ID: >>> 195110554Rsep@84.51.32.26..CSeq: 1 INVITE..Contact: <sip:90 >>> 2247654321(a)84.51.32.26>..Supported: >>> em,100rel,timer,replaces,path..Allow: >>> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDAT >>> E..Session-Expires: 3600..Min-SE: 90..User-Agent: >>> Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.200.371..Content-Type: >>> application/sdp..Content-Length: 244.. >>> ..v=0..o=AudiocodesGW 484853 895184 IN IP4 >>> 84.51.32.26..s=Phone-Call..c=IN IP4 84.51.32.26..t=0 0..m=audio 4010 >>> RTP/AVP 0 8 96..a=rtpmap:0 pcmu/8000..a= >>> rtpmap:8 pcma/8000..a=rtpmap:96 telephone-event/8000..a=fmtp:96 >>> 0-15..a=ptime:20..a=sendrecv.. >>> # >>> U 84.51.32.8:5060 -> 84.51.32.26:5060 >>> SIP/2.0 403 ONLY REGISTERED USERS......Via: SIP/2.0/UDP >>> 84.51.32.26;branch=z9hG4bKacGtOiLfA..From: "902247654321" >>> <sip:902247654321@gi.com.tr>;tag=1c572 >>> 128494..To: >>> <sip:02124440111@gi.com.tr;user=phone>;tag=26a82380ee921ee699cdfa26683b3165.a215..Call-ID: >>> 195110554Rsep@84.51.32.26..CSeq: 1 INVITE..Server >>> : Sip EXpress router (0.9.3 (i386/freebsd))..Content-Length: >>> 0..Warning: 392 84.51.32.8:5060 "Noisy feedback tells: pid=70414 >>> req_src_ip=84.51.32.26 re >>> q_src_port=5060 in_uri=sip:02124440111@gi.com.tr;user=phone >>> out_uri=sip:02124440111@gi.com.tr;user=phone via_cnt==1".... >>> # >>> U 84.51.32.26:5060 -> 84.51.32.8:5060 >>> ACK sip:02124440111@gi.com.tr;user=phone SIP/2.0..Via: SIP/2.0/UDP >>> 84.51.32.26;branch=z9hG4bKacGtOiLfA..Max-Forwards: 70..From: >>> "902247654321" <sip:9022 >>> 47654321(a)gi.com.tr>;tag=1c572128494..To: >>> <sip:02124440111@gi.com.tr;user=phone>;tag=26a82380ee921ee699cdfa26683b3165.a215..Call-ID: >>> 195110554Rsep(a)84.51. >>> 32.26..CSeq: 1 ACK..Contact: >>> <sip:902247654321@84.51.32.26>..Supported: >>> em,timer,replaces,path..Allow: >>> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRA >>> CK,REFER,INFO,SUBSCRIBE,UPDATE..User-Agent: >>> Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.200.371..Content-Length: 0.... >>> >>> ################ SER LOG ########### >>> 2(70414) val2str(): converting 84.51.32.26, 11 >>> 2(70414) PG[217] str2valp got string udp >>> 2(70414) PG[217] str2valp got string "90224765321" >>> 2(70414) NOT TRUSTED IP 2(70414) parse_headers: flags=4 >>> 2(70414) end of header reached, state=9 >>> 2(70414) DEBUG: get_hdr_field: <To> [40]; >>> uri=[sip:02124440111@gi.com.tr;user=phone] >>> 2(70414) DEBUG: to body [<sip:02124440111@gi.com.tr;user=phone> >>> >>> >>> Best Regards, >>> Hakan. >>> >>> >>> >>> >>> >>> >>> ----- Original Message ----- From: "Miklos Tirpak" <miklos(a)iptel.org> >>> To: "Hakan YASTI" <hakanyasti(a)gmail.com> >>> Cc: <serusers(a)lists.iptel.org> >>> Sent: Thursday, March 02, 2006 11:38 AM >>> Subject: Re: [Serusers] permission module >>> >>> >>>> Hi Hakan, >>>> >>>> Does the From header really look like this??? >>>> From: "902247654321" ;tag=1c681629589 >>>> >>>> It is incorrect, and of course the To and Contact headers are also >>>> incorrect. >>>> >>>> Miklos >>>> >>>> Hakan YASTI wrote: >>>> >>>>> Hi, >>>>> I am trying to use permissions module. At configuration side >>>>> everything seems ok. when an invite comes to ser, gets the source >>>>> ip of the invite and select a query from the trusted table. But >>>>> every time, when correlates the from_pattern and proto, they >>>>> mismatch ( I SUPPOSE ). Is there any trick at from_pattern ? I have >>>>> tried so many values like >>>>> .* , sip:.*,empty,^sip:.* and etc. Here is my ser.cfg, logs. >>>>> >>>>> Hakan. >>>>> >>>>> loadmodule "/usr/local/lib/ser/modules/permissions.so" >>>>> # -- permissions parameters -- # >>>>> >>>>> modparam("permissions", "db_url", >>>>> "postgres://xxxxx:xxxx@ip_address/xxxxx") >>>>> modparam("permissions", "db_mode", 0) >>>>> modparam("permissions", "trusted_table", "trusted") >>>>> modparam("permissions", "source_col", "src_ip") >>>>> modparam("permissions", "proto_col", "proto") >>>>> modparam("permissions", "from_col", "from_pattern") >>>>> >>>>> if (method=="INVITE") { >>>>> if (!allow_trusted()) { >>>>> log("THE IP ADDRESS IS NOT ALLOWED"); >>>>> sl_send_reply("403","ONLY REGISTERED USERS ALLOWED "); >>>>> break; >>>>> } >>>>> >>>>> >>>>> >>>>> ########### SER LOG ############# >>>>> >>>>> 2(69289) SIP Request: >>>>> 2(69289) method: >>>>> 2(69289) uri: >>>>> 2(69289) version: >>>>> 2(69289) parse_headers: flags=1 >>>>> 2(69289) Found param type 232, = ; state=16 >>>>> 2(69289) end of header reached, state=5 >>>>> 2(69289) parse_headers: Via found, flags=1 >>>>> 2(69289) parse_headers: this is the first via >>>>> 2(69289) After parse_msg... >>>>> 2(69289) preparing to run routing scripts... >>>>> 2(69289) parse_headers: flags=128 >>>>> 2(69289) DEBUG:maxfwd:is_maxfwd_present: value = 70 >>>>> 2(69289) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 >>>>> 2(69289) parse_headers: flags=8 >>>>> 2(69289) DEBUG: add_param: tag=1c681629589 >>>>> 2(69289) end of header reached, state=29 >>>>> 2(69289) grep_sock_info - checking if host==us: 9==10 && >>>>> [gi.com.tr] == [84.51.32.8] >>>>> 2(69289) grep_sock_info - checking if port 5060 matches port 5060 >>>>> 2(69289) grep_sock_info - checking if host==us: 9==9 && [gi.com.tr] >>>>> == [127.0.0.1] >>>>> 2(69289) grep_sock_info - checking if port 5060 matches port 5060 >>>>> 2(69289) grep_sock_info - checking if host==us: 9==10 && >>>>> [gi.com.tr] == [84.51.32.8] >>>>> 2(69289) grep_sock_info - checking if port 5060 matches port 5060 >>>>> 2(69289) grep_sock_info - checking if host==us: 9==9 && [gi.com.tr] >>>>> == [127.0.0.1] >>>>> 2(69289) grep_sock_info - checking if port 5060 matches port 5060 >>>>> 2(69289) val2str(): converting 84.51.32.26, 11 >>>>> 2(69289) PG[217] str2valp got string udp >>>>> 2(69289) PG[217] str2valp got string sip:90224765321@gi.com.tr >>>>> 2(69289) NOT TRUSTED IP 2(69289) parse_headers: flags=4 >>>>> 2(69289) end of header reached, state=9 >>>>> 2(69289) DEBUG: get_hdr_field: [40]; >>>>> uri=[sip:02124440111@gi.com.tr;user=phone] >>>>> 2(69289) DEBUG: to body [ >>>>> ] >>>>> 2(69289) parse_headers: flags=-1 >>>>> 2(69289) get_hdr_field: cseq : <1> >>>>> 2(69289) DEBUG: get_hdr_body : content_length=267 >>>>> 2(69289) found end of header >>>>> 2(69289) check_via_address(84.51.32.26, 84.51.32.26, 1) >>>>> 2(69289) DEBUG:destroy_avp_list: destroying list 0x0 >>>>> 2(69289) receive_msg: cleaning up >>>>> 3(69290) SIP Request: >>>>> 3(69290) method: >>>>> 3(69290) uri: >>>>> 3(69290) version: >>>>> 3(69290) parse_headers: flags=1 >>>>> 3(69290) Found param type 232, = ; state=16 >>>>> 3(69290) end of header reached, state=5 >>>>> 3(69290) parse_headers: Via found, flags=1 >>>>> 3(69290) parse_headers: this is the first via >>>>> 3(69290) After parse_msg... >>>>> 3(69290) preparing to run routing scripts... >>>>> 3(69290) parse_headers: flags=4 >>>>> 3(69290) DEBUG: add_param: tag=26a82380ee921ee699cdfa26683b3165.bcf6 >>>>> 3(69290) end of header reached, state=29 >>>>> 3(69290) DEBUG: get_hdr_field: [82]; >>>>> uri=[sip:02124440111@gi.com.tr;user=phone] >>>>> 3(69290) DEBUG: to body [] >>>>> 3(69290) DEBUG: sl_filter_ACK : local ACK found -> dropping it! >>>>> 3(69290) DEBUG:destroy_avp_list: destroying list 0x0 >>>>> 3(69290) receive_msg: cleaning up >>>>> >>>>> >>>>> >>>>> >>>>> ######### NGREP LOGS ################ >>>>> >>>>> U 84.51.32.26:5060 -> 84.51.32.8:5060 >>>>> INVITE sip:02124440111@gi.com.tr;user=phone SIP/2.0. >>>>> Via: SIP/2.0/UDP 84.51.32.26;branch=z9hG4bKaccQkWrLg. >>>>> Max-Forwards: 70. >>>>> From: "902247654321" ;tag=1c681629589. >>>>> To: . >>>>> Call-ID: *395527579dNhC(a)84.51.32.26.* >>>>> <mailto:395527579dNhC@84.51.32.26.> >>>>> CSeq: 1 INVITE. >>>>> Contact: . >>>>> Supported: em,100rel,timer,replaces,path. >>>>> Allow: >>>>> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. >>>>> >>>>> Session-Expires: 3600. >>>>> Min-SE: 90. >>>>> User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.200.371. >>>>> Content-Type: application/sdp. >>>>> Content-Length: 267. >>>>> . >>>>> v=0. >>>>> o=AudiocodesGW 440639 653740 IN IP4 84.51.32.26. >>>>> s=Phone-Call. >>>>> c=IN IP4 84.51.32.26. >>>>> t=0 0. >>>>> m=audio 4010 RTP/AVP 18 8 96. >>>>> a=rtpmap:18 g729/8000. >>>>> a=fmtp:18 annexb=no. >>>>> a=rtpmap:8 pcma/8000. >>>>> a=rtpmap:96 telephone-event/8000. >>>>> a=fmtp:96 0-15. >>>>> a=ptime:40. >>>>> a=sendrecv. >>>>> >>>>> # >>>>> U 84.51.32.8:5060 -> 84.51.32.26:5060 >>>>> SIP/2.0 403 ONLY REGISTERED USERS..... >>>>> Via: SIP/2.0/UDP 84.51.32.26;branch=z9hG4bKaccQkWrLg. >>>>> From: "902247654321" ;tag=1c681629589. >>>>> To: ;tag=26a82380ee921ee699cdfa26683b3165.bcf6. >>>>> Call-ID: *395527579dNhC(a)84.51.32.26.* >>>>> <mailto:395527579dNhC@84.51.32.26.> >>>>> CSeq: 1 INVITE. >>>>> Server: Sip EXpress router (0.9.3 (i386/freebsd)). >>>>> Content-Length: 0. >>>>> Warning: 392 84.51.32.8:5060 "Noisy feedback tells: pid=69289 >>>>> req_src_ip=84.51.32.26 req_src_port=5060 >>>>> in_uri=sip:02124440111@gi.com.tr;user=phone >>>>> out_uri=sip:02124440111@gi.com.tr;user=phone via_cnt==1". >>>>> . >>>>> >>>>> # >>>>> U 84.51.32.26:5060 -> 84.51.32.8:5060 >>>>> ACK sip:02124440111@gi.com.tr;user=phone SIP/2.0. >>>>> Via: SIP/2.0/UDP 84.51.32.26;branch=z9hG4bKaccQkWrLg. >>>>> Max-Forwards: 70. >>>>> From: "902247654321" ;tag=1c681629589. >>>>> To: ;tag=26a82380ee921ee699cdfa26683b3165.bcf6. >>>>> Call-ID: *395527579dNhC(a)84.51.32.26.* >>>>> <mailto:395527579dNhC@84.51.32.26.> >>>>> CSeq: 1 ACK. >>>>> Contact: . >>>>> Supported: em,timer,replaces,path. >>>>> Allow: >>>>> REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. >>>>> >>>>> User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.200.371. >>>>> Content-Length: 0. >>>>> >>>>> ################################# >>>>> Tried another value for from_pattern... >>>>> >>>>> 3(69290) val2str(): converting 84.51.32.26, 11 >>>>> 3(69290) PG[217] str2valp got string udp >>>>> 3(69290) PG[217] str2valp got string "90224765321" >>>>> >>>>> >>>>> >>>>> select * from trusted; >>>>> src_ip | proto | from_pattern >>>>> -------------+-------+--------------- >>>>> 84.51.32.25 | udp | "90224765321" >>>>> 84.51.32.26 | udp | "90224765321" >>>>> (2 rows) >>>>> . >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> >>>>> >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> serusers(a)lists.iptel.org >>>>> http://lists.iptel.org/mailman/listinfo/serusers

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[Users] Route found, but not able to forward call
by Florian Fainelli 03 Mar '06

03 Mar '06

Hello all, I am progressing in my dynamic routing with lcr. I am able to make my calls being routed to a given gateway, but they are not forwarded at all, you will find below the error messages I get : 12(13607) SIP Request: 12(13607) method: <CANCEL> 12(13607) uri: <sip:60957512@213.161.196.133> 12(13607) version: <SIP/2.0> 12(13607) parse_headers: flags=2 12(13607) header field type 8, name=<Max-Forwards>, body=<10> 12(13607) Found param type 232, <branch> = <1>; state=16 12(13607) end of header reached, state=5 12(13607) parse_headers: Via found, flags=2 12(13607) parse_headers: this is the first via 12(13607) header field type 1, name=<Via>, body=<SIP/2.0/UDP 157.159.80.57:5060;branch=1> 12(13607) first via: <SIP/2.0/UDP> <157.159.80.57:5060(5060)>12(13607) ;<>12(13607) 12(13607) exiting parse_msg 12(13607) After parse_msg... 12(13607) preparing to run routing scripts... 12(13607) DEBUG:maxfwd:is_maxfwd_present: value = 10 12(13607) parse_headers: flags=200 12(13607) header field type 4, name=<From>, body=<<sip:157.159.80.57>;tag=044864755209883246007508179717> 12(13607) DEBUG:parse_to:end of header reached, state=9 12(13607) DEBUG: get_hdr_field: <To> [32]; uri=[sip:60957512@213.161.196.133] 12(13607) DEBUG: to body [<sip:60957512@213.161.196.133> ] 12(13607) header field type 3, name=<To>, body=<<sip:60957512@213.161.196.133>> 12(13607) header field type 7, name=<Contact>, body=<<sip:157.159.80.57>> 12(13607) header field type 26, name=<User-Agent>, body=<TELES.VoIPBOX 10.0 #302> 12(13607) header field type 6, name=<Call-ID>, body=<000688355273195645908699556113(a)157.159.80.57> 12(13607) get_hdr_field: cseq <CSeq>: <1> <CANCEL> 12(13607) header field type 5, name=<CSeq>, body=<1 CANCEL> 12(13607) header field type 19, name=<Allow>, body=<INVITE,ACK,CANCEL,BYE,UPDATE,REGISTER> 12(13607) DEBUG: get_hdr_body : content_length=0 12(13607) header field type 12, name=<Content-Length>, body=<0> 12(13607) found end of header 12(13607) find_first_route: No Route headers found 12(13607) loose_route: There is no Route HF 12(13607) parsed uri: type=1 user=<60957512>(8) passwd=<>(0) host=<213.161.196.133>(15) port=<>(0): 0 params=<>(0) headers=<>(0) 12(13607) uri params: transport=<>, val=<>, proto=0 12(13607) user-param=<>, val=<> 12(13607) method=<>, val=<> 12(13607) ttl=<>, val=<> 12(13607) maddr=<>, val=<> 12(13607) lr=<> 12(13607) DEBUG: add_param: tag=044864755209883246007508179717 12(13607) DEBUG:parse_to:end of header reached, state=29 12(13607) load_gws(): DEBUG: Added gw_uri_avp <sip:@82.138.76.138:5060;transport=tcp> 12(13607) DEBUG:destroy_avp_list: destroying list 0xb5c01f10 12(13607) receive_msg: cleaning up Thanks in advance for your answers. -- Cordialement, Florian Fainelli --------------------------------------- 5, rue Charles Fourier Chambre 1511 91011 Evry http://www.alphacore.net (+33) 01 60 76 64 86 (+33) 06 09 02 64 95 ---------------------------------------- Institut National des Télécomnications http://www.int-evry.fr/telecomint ----------------------------------------

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[Serusers] serctl PID file exists error with 0.9.7-pre1
by Steve Blair 03 Mar '06

03 Mar '06

starting SER 0.9.7-pre1 with serctl results in the PID file exists error. SER will start with the command ser -d -E -f <config file>. Any thoughts on what might be wrong? Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:blairs@net.isc.upenn.edu

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[Users] pseudovaribles in append_to_reply() not working
by Jose Antonio Garvayo 03 Mar '06

03 Mar '06

Hi, I am experiencing a problem with pseudovariables in append_to_reply(), seems like they are not being replaced with the real value. In openser.cfg, I am using this: append_to_reply("P-Hint: $Ri : $Rp \r\n"); But I am receiving this: ... P-Hint: $Ri : $Rp Server: OpenSer (1.0.1 (i386/linux)) ... I tried also the same in a append_hf() with the same result, the pseudovariables are not working. I am using openser 1.0.1 in linux, does anyone have any clue on this? Thanks a lot

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Re: [Serusers] how to let iptel ser forward an invite message with route header?
by fang tian 03 Mar '06

03 Mar '06

Ladislav, the problem is ,if I didn't add these two route headers,the application server will do a DNS lookup,which is more than 30 seconds. I also want to make Iptel ser looks like a S-CSCF which defined in 3GPP standard. thanks Fang tian. On 3/3/06, Ladislav Andel <ladia6(a)centrum.cz> wrote: > > As far as I know you don't need to add any route header in this case. On > the top of SIP message is VIA header which says which > way your application server should answer back.. > > Route header is used when you have more proxies in the way where the > message you want to route through > > from RFC 3261: > > The Route header field is used to force routing for a request through > the listed set of proxies. Examples of the use of the Route header > field are in Section 16.12.1. > > Example: > > Route: <sip:bigbox3.site3.atlanta.com;lr>, > <sip:server10.biloxi.com;lr> > > Can anybody confirm? > > Ladislav > > fang tian wrote: > > Hi , > > I am a new user of iptel ser,I want to make this configuration > > > > UE1,2,3,4<---------->IPtel ser > > <---------------------------------->application server , > > > > the UEs send out invite request to iptel ser first ,I want to make > > iptel ser forward this request to application server with two route > > header added: > > route:application server :5060 > > route:iptel ser :5060 > > > > so the application server can send this request back to iptel ser. > > > > your kindly help is appreciated ,thanks > > > > RGS > > tian. > > > > > > > > -- > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > -- Best regards Fang Tian.

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[Serusers] how to let iptel ser forward an invite message with route header?
by fang tian 03 Mar '06

03 Mar '06

Hi , I am a new user of iptel ser,I want to make this configuration UE1,2,3,4<---------->IPtel ser <---------------------------------->application server , the UEs send out invite request to iptel ser first ,I want to make iptel ser forward this request to application server with two route header added: route:application server :5060 route:iptel ser :5060 so the application server can send this request back to iptel ser. your kindly help is appreciated ,thanks RGS tian. --

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[Serusers] SEMS + G.729
by Tomasz Stafin 03 Mar '06

03 Mar '06

Hi again I have another problem with SEMS. It works fine with codec G.711 but I'd like to use codec G.729. Of course I'd like to buy a licence but I don't know how to do it and if SEMS work with codec G.729. If anyone use this codec with SEMS please help me to configure SEMS with it. Thanks for any help. Swaper

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[Serusers] Visual Message Waiting indicator
by tdeillon＠cri74.org 03 Mar '06

03 Mar '06

Dear all, I use: - ser as proxy, registrar and redirect server. - asterisk as gateway and voicemail server. But my problem is that I don't have the "Visual Message Waiting indicator" on my phones. Do you have a idea for me ? Best regards, Thomas

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Re: [Serusers] rtpproxy
by Andrey Kouprianov 03 Mar '06

03 Mar '06

Sorry guy for the russian language here.... Vitaliy, privet! Ya iz Thailanda... Tochnee kak, zhivu i rabotayu tut (po krainei mere poka), a voobshe iz Moskvy. Zanimayus' razrabotkoi SIP clienta dlya E-learning na Jave. Poka dlya PC i laptopov, a potom nado vayat' analogichnuyu progu na PDA i mobil'niye telefony. Priyatno poznakomitsya :)) Andrey. On 3/3/06, Vitaly Nikolaev <vnikolaev(a)intermedia.net> wrote: > Ahh... :) netuda :) > > A ti otkuda takoi znauchii kstati ? :) > > > -----Original Message----- > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On > Behalf Of Andrey Kouprianov > Sent: Thursday, March 02, 2006 12:09 PM > To: serusers(a)lists.iptel.org > Subject: Re: [Serusers] rtpproxy > > Im not sure about it, cause I never had chance to play with PSTN > numbers properly. There's something about DID's on page 58 of the > document. > > On 3/3/06, roger leszczynski <rogerles(a)gmail.com> wrote: > > i found it, thanks, now the question is, say i have 100 concurrent > calls, > > and i want to update the configuration with a new DID, how can i avoid > > restarting SER to make these changes take places....has anyone made a > > similar module to reload in asterisk? Or how can i go about using a > > database to implement routing instead? > > > > > > On 3/2/06, Andrey Kouprianov <andrey.kouprianov(a)gmail.com > wrote: > > > > > would "getting started" document from www.onsip.org help? they have a > > good explanation on how to configure SER with RTPProxy... > > > > On 3/2/06, roger leszczynski <rogerles(a)gmail.com> wrote: > > > Anyone use the rtp proxy module yet? I am looking to proxy rtp of > my > > > carrier who sends out calls to me, so if anyone has example > configurations > > > that work i would appreciate it! > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > _______________________________________________ > > > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Re: SER manual
by Andrey Kouprianov 03 Mar '06

03 Mar '06

Sign in first. Then right in the middle of the page there will be Top Downloads Section. Click on "Issue 50 - Getting Started" (not "Issue 5.0 ... "). On the new page, click on "Issue 50 - Getting Started" link located near the floppy disk icon. On 3/3/06, Marcio Sousa <mcsousa(a)av.it.pt> wrote: > Hi, > I´ve been searching in www.onsip.org for getting started Issue 5 PDF > document, but I didn´t find any pdf doc. Can you please precise where > can I find such doc, or any place where I can get information about SER > scripts, beside iptel.org? > Thanks. >

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[Serusers] where can i find documents for ser.cfg.
by wu yaqian 03 Mar '06

03 Mar '06

Hello,erery one I want to write ser.cfg myself.where can i find some documents about this subject. thanks a lot

2 1

[Serusers] changing position without closing the current session
by Marcio Sousa 03 Mar '06

03 Mar '06

Hi, When a convertation between 2 clients starts, is it possible to change the position of one of the clients, without closing their current session (mobility), using SIP signaling? Could you give me any idea how to make this possible? Thanks .

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[Serusers] TCP connect and other related stuff problems
by Christian Thomas 02 Mar '06

02 Mar '06

Hi everyboy, I have a problem when my SER 0.9.4 has to connect to a carrier thru tcp transport. I'm working in stateful mode, of course, and I use t_relay_to_tcp(ipcarrier,port) when I can detect that the carrier GW wants to speak with SER thru TCP. It works not very well as there is the following errors when the carrier GW send a 200 OK to SER after the INVITE procedure : tcp_blocking_connect : timeout(10) tcpconn_connect : connect failed tcp_send: connect failed tcp_send failed msg_send : tcp_send failed t_forward_nonack : sending request failed The UAS which sent the INVITE receive a 477 : next hop error Is it a timeout problem ? If so, how to resolve it? How to increase this timeout for TCP? Thanks a lot Christian

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[Serusers] (no subject)
by Richard C. Thompson 02 Mar '06

02 Mar '06

Hi All I have a problem auto starting my new mediaproxy server. I'm using mediaproxy 1.4.2 with mediaproxy.ini file. I have /etc/init.d/mediaproxy in place and the symlink in rc3.d I noticed that the start file is the same as the one I have for mediaproxy 1.2.1 and yet the Changelog says it should be different. Any ideas on what I should look at? Rick t

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[Serusers] sending INVITES
by Richard C. Thompson 02 Mar '06

02 Mar '06

All I ran into a small problem with our SER server sending calls to a Quintum CRSP or Call Relay Server. Here is what I received back from the Quintum software group. The SIP device is our SER server. The Quintum CRSP is used to take inbound SIP calls and sends them out as H323 calls to a terminating PSTN gateway. SIP device makes call #1 to CRSP Call #1 gets alerting from H323 The same SIP device makes call #2 with same SIP call-ID, same ANI, Same DNIS Call #2 Fails CRSP frees the call #1 since it has the same call id Call #1 receives connect from H323 side System hangs They are in the process of changing their software to accommodate this SER issue but have advised me to contact the makers of the SIP device and let them know they are not following the SIP standard. I believe the SER is send a second "INVITE" because the CRSP didn't reply fast enough, or it is sending a re-"INVITE" after an "INVITE" timeout, which could be my bad. Rick T

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RE: [Serusers] reload
by Vitaly Nikolaev 02 Mar '06

02 Mar '06

Waw... what did u use as call generator? What was the server and what was call rate ? (500 sim calls is not so important as how many calls per second, I can calculate it if u tell me what was call duration :)) What was cpu load, and... what was SER response time (how much time it take to serve one invite when you have 500 sim calls (up and down)) -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Kapil Dhawan Sent: Thursday, March 02, 2006 1:04 PM To: tobias.lindgren(a)ip-only.se; rogerles(a)gmail.com Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] reload Hi Tested it with around 500 Simultaneous calls. Runs perfectly. >From: Tobias Lindgren <tobias.lindgren(a)ip-only.se> >To: roger leszczynski <rogerles(a)gmail.com> >CC: serusers(a)lists.iptel.org >Subject: Re: [Serusers] reload >Date: Thu, 02 Mar 2006 18:16:16 +0100 > >Hi, > >the simples way to implement a database for routing is installing a >MYSQL-db and create a table which holds your number and the routing >destination for that number (ip-address). Then you create a perl/php/C >script that queries that database using the exec module in ser. Or >better yet, create a ser-module of your own that makes those queries. > >However, I'm not really sure how well this scales when you are using >exec and start loading your server, but if your scripts and your >database are well created, I guess you should be fine for while. > >Br, >Tobias > >roger leszczynski said the following on 2006-03-02 18:01: > > say i have 100 concurrent calls, and i want to update the configuration > > with a new DID, how can i avoid restarting SER to make these changes > > take places....has anyone made a similar module to reload in asterisk? > > Or how can i go about using a database to implement routing instead? > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers _________________________________________________________________ Sexy, sultry, sensuous. - see why Bipasha Basu is all that and more. Try MSN Search http://server1.msn.co.in/Profile/bipashabasu.asp _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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RE: [Serusers] rtpproxy
by Vitaly Nikolaev 02 Mar '06

02 Mar '06

Roger, Most of the time SER has only basic configuration, with static routing, all dynamic stuff will do b2bua or via ser modules like lcr, or via aliases that keep all information in mysql databases. Let say you have customers registered on ser and you have incoming DID that all saved in mysql and ser use lookup(aliases) will route did to designated customer, if u want to change it you change in mysql without restart of ser Good luck -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Andrey Kouprianov Sent: Thursday, March 02, 2006 12:09 PM To: serusers(a)lists.iptel.org Subject: Re: [Serusers] rtpproxy Im not sure about it, cause I never had chance to play with PSTN numbers properly. There's something about DID's on page 58 of the document. On 3/3/06, roger leszczynski <rogerles(a)gmail.com> wrote: > i found it, thanks, now the question is, say i have 100 concurrent calls, > and i want to update the configuration with a new DID, how can i avoid > restarting SER to make these changes take places....has anyone made a > similar module to reload in asterisk? Or how can i go about using a > database to implement routing instead? > > > On 3/2/06, Andrey Kouprianov <andrey.kouprianov(a)gmail.com > wrote: > > > would "getting started" document from www.onsip.org help? they have a > good explanation on how to configure SER with RTPProxy... > > On 3/2/06, roger leszczynski <rogerles(a)gmail.com> wrote: > > Anyone use the rtp proxy module yet? I am looking to proxy rtp of my > > carrier who sends out calls to me, so if anyone has example configurations > > that work i would appreciate it! > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > _______________________________________________ > > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] reload
by roger leszczynski 02 Mar '06

02 Mar '06

say i have 100 concurrent calls, and i want to update the configuration with a new DID, how can i avoid restarting SER to make these changes take places....has anyone made a similar module to reload in asterisk? Or how can i go about using a database to implement routing instead?

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Re: [Serusers] rtpproxy
by Andrey Kouprianov 02 Mar '06

02 Mar '06

Im not sure about it, cause I never had chance to play with PSTN numbers properly. There's something about DID's on page 58 of the document. On 3/3/06, roger leszczynski <rogerles(a)gmail.com> wrote: > i found it, thanks, now the question is, say i have 100 concurrent calls, > and i want to update the configuration with a new DID, how can i avoid > restarting SER to make these changes take places....has anyone made a > similar module to reload in asterisk? Or how can i go about using a > database to implement routing instead? > > > On 3/2/06, Andrey Kouprianov <andrey.kouprianov(a)gmail.com > wrote: > > > would "getting started" document from www.onsip.org help? they have a > good explanation on how to configure SER with RTPProxy... > > On 3/2/06, roger leszczynski <rogerles(a)gmail.com> wrote: > > Anyone use the rtp proxy module yet? I am looking to proxy rtp of my > > carrier who sends out calls to me, so if anyone has example configurations > > that work i would appreciate it! > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > _______________________________________________ > > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > >

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[Serusers] rtpproxy
by roger leszczynski 02 Mar '06

02 Mar '06

Anyone use the rtp proxy module yet? I am looking to proxy rtp of my carrier who sends out calls to me, so if anyone has example configurations that work i would appreciate it!

2 1

[Serusers] Redirect a sip outbound requests to a sip proxy
by hgaillac-sip＠yahoo.fr 02 Mar '06

02 Mar '06

Hi all, Is there a solution to solve this ? ASTERISK 1.2.4 || Internet===SER/OPENSER=====Nat==[private net] || sip agents rtpproxy/mediaproxy Sip agents use SER/OPENSER as an outbound sip proxy and asterisk as a registar server, pbx functions, ... SER/OPENSER look for domains in URI. if domains are handled by SER/OPENSER so requests are forwarded to ASTERISK. If domain in URI is not handled by SER/OPENSER and/or ASTERISK I wish the request is sent to Asterisk before to send back it to SER/OPENSER. in other word i wish to define SER/OPENSER server as oubound proxy for Astersik. Is it Possible ? Harry ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

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[Serusers] serctl problem
by frankie chiang 02 Mar '06

02 Mar '06

Dear all, Please let me know what happen. Just create a new database by ser_mysql.sh script. # serctl add 200 200 200(a)test.com MySql password: error: 400; check if you use aliases in SER Best Regard, __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] using SER as a frontend for multiple asterisk servers?
by Roy Sigurd Karlsbakk 02 Mar '06

02 Mar '06

hi does anyone have a source of documentation that can help me setting up the following? one SER box (with a potential additional box for failover) should - Accept REGISTER and so on and just balance these over n asterisk boxes in behind. No authentication should be done on the SER box. - Accept INVITE and distribute this linearly to n asterisk boxes behind, starting with box 1, allow up to n calls to each box (each box having n PRI/E1s). - Accept INVITE from the asterisk boxes and forward these to the client - All RTP traffic should be proxied locally All help appreciated :) roy

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Re: [Serusers] classkit.so module php5
by Kim Culhan 02 Mar '06

02 Mar '06

On Wed, March 1, 2006 5:01 pm, Ofer Zaarur said: > I have problem getting classkit module to set serweb correctly using > php5. it looks like this issue had been raise before. please let me know > where I can get it. You may find some problems running serweb with PHP5, much easier to run PHP4 as it has classkit. If you really want to try here is an example for building classkit at /usr/local/src: Get the source for php5 and untar to get: /usr/local/src/php5/php-5.1.2 Get the source for classkit from: http://pecl.php.net/package/classkit cd /usr/local/src/php5/php-5.1.2/ext and untar the source from the pecl web site for classkit in the directory above and rename the directory from classkit-0.4 to classkit. Then cd .. to the top level of the php-5.1.2 build system and run: phpize && ./configure --with-classkit cd to ext/classkit and you will now find a complete build tree run configure and make and you will get classkit.dsp, classkit.la, classkit.lo When you type make install it will install classkit.so in /usr/local/lib/php/20050922 The shared library version with the .so suffix is created in the above step. Hope this helps -kim -- w8hdkim(a)gmail.com

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[Serusers] permission module
by Hakan YASTI 02 Mar '06

02 Mar '06

Hi, I am trying to use permissions module. At configuration side everything seems ok. when an invite comes to ser, gets the source ip of the invite and select a query from the trusted table. But every time, when correlates the from_pattern and proto, they mismatch ( I SUPPOSE ). Is there any trick at from_pattern ? I have tried so many values like .* , sip:.*,empty,^sip:.* and etc. Here is my ser.cfg, logs. Hakan. loadmodule "/usr/local/lib/ser/modules/permissions.so" # -- permissions parameters -- # modparam("permissions", "db_url", "postgres://xxxxx:xxxx@ip_address/xxxxx") modparam("permissions", "db_mode", 0) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "source_col", "src_ip") modparam("permissions", "proto_col", "proto") modparam("permissions", "from_col", "from_pattern") if (method=="INVITE") { if (!allow_trusted()) { log("THE IP ADDRESS IS NOT ALLOWED"); sl_send_reply("403","ONLY REGISTERED USERS ALLOWED "); break; } ########### SER LOG ############# 2(69289) SIP Request: 2(69289) method: 2(69289) uri: 2(69289) version: 2(69289) parse_headers: flags=1 2(69289) Found param type 232, = ; state=16 2(69289) end of header reached, state=5 2(69289) parse_headers: Via found, flags=1 2(69289) parse_headers: this is the first via 2(69289) After parse_msg... 2(69289) preparing to run routing scripts... 2(69289) parse_headers: flags=128 2(69289) DEBUG:maxfwd:is_maxfwd_present: value = 70 2(69289) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 2(69289) parse_headers: flags=8 2(69289) DEBUG: add_param: tag=1c681629589 2(69289) end of header reached, state=29 2(69289) grep_sock_info - checking if host==us: 9==10 && [gi.com.tr] == [84.51.32.8] 2(69289) grep_sock_info - checking if port 5060 matches port 5060 2(69289) grep_sock_info - checking if host==us: 9==9 && [gi.com.tr] == [127.0.0.1] 2(69289) grep_sock_info - checking if port 5060 matches port 5060 2(69289) grep_sock_info - checking if host==us: 9==10 && [gi.com.tr] == [84.51.32.8] 2(69289) grep_sock_info - checking if port 5060 matches port 5060 2(69289) grep_sock_info - checking if host==us: 9==9 && [gi.com.tr] == [127.0.0.1] 2(69289) grep_sock_info - checking if port 5060 matches port 5060 2(69289) val2str(): converting 84.51.32.26, 11 2(69289) PG[217] str2valp got string udp 2(69289) PG[217] str2valp got string sip:90224765321@gi.com.tr 2(69289) NOT TRUSTED IP 2(69289) parse_headers: flags=4 2(69289) end of header reached, state=9 2(69289) DEBUG: get_hdr_field: [40]; uri=[sip:02124440111@gi.com.tr;user=phone] 2(69289) DEBUG: to body [ ] 2(69289) parse_headers: flags=-1 2(69289) get_hdr_field: cseq : <1> 2(69289) DEBUG: get_hdr_body : content_length=267 2(69289) found end of header 2(69289) check_via_address(84.51.32.26, 84.51.32.26, 1) 2(69289) DEBUG:destroy_avp_list: destroying list 0x0 2(69289) receive_msg: cleaning up 3(69290) SIP Request: 3(69290) method: 3(69290) uri: 3(69290) version: 3(69290) parse_headers: flags=1 3(69290) Found param type 232, = ; state=16 3(69290) end of header reached, state=5 3(69290) parse_headers: Via found, flags=1 3(69290) parse_headers: this is the first via 3(69290) After parse_msg... 3(69290) preparing to run routing scripts... 3(69290) parse_headers: flags=4 3(69290) DEBUG: add_param: tag=26a82380ee921ee699cdfa26683b3165.bcf6 3(69290) end of header reached, state=29 3(69290) DEBUG: get_hdr_field: [82]; uri=[sip:02124440111@gi.com.tr;user=phone] 3(69290) DEBUG: to body [] 3(69290) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 3(69290) DEBUG:destroy_avp_list: destroying list 0x0 3(69290) receive_msg: cleaning up ######### NGREP LOGS ################ U 84.51.32.26:5060 -> 84.51.32.8:5060 INVITE sip:02124440111@gi.com.tr;user=phone SIP/2.0. Via: SIP/2.0/UDP 84.51.32.26;branch=z9hG4bKaccQkWrLg. Max-Forwards: 70. From: "902247654321" ;tag=1c681629589. To: . Call-ID: 395527579dNhC(a)84.51.32.26. CSeq: 1 INVITE. Contact: . Supported: em,100rel,timer,replaces,path. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. Session-Expires: 3600. Min-SE: 90. User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.200.371. Content-Type: application/sdp. Content-Length: 267. . v=0. o=AudiocodesGW 440639 653740 IN IP4 84.51.32.26. s=Phone-Call. c=IN IP4 84.51.32.26. t=0 0. m=audio 4010 RTP/AVP 18 8 96. a=rtpmap:18 g729/8000. a=fmtp:18 annexb=no. a=rtpmap:8 pcma/8000. a=rtpmap:96 telephone-event/8000. a=fmtp:96 0-15. a=ptime:40. a=sendrecv. # U 84.51.32.8:5060 -> 84.51.32.26:5060 SIP/2.0 403 ONLY REGISTERED USERS..... Via: SIP/2.0/UDP 84.51.32.26;branch=z9hG4bKaccQkWrLg. From: "902247654321" ;tag=1c681629589. To: ;tag=26a82380ee921ee699cdfa26683b3165.bcf6. Call-ID: 395527579dNhC(a)84.51.32.26. CSeq: 1 INVITE. Server: Sip EXpress router (0.9.3 (i386/freebsd)). Content-Length: 0. Warning: 392 84.51.32.8:5060 "Noisy feedback tells: pid=69289 req_src_ip=84.51.32.26 req_src_port=5060 in_uri=sip:02124440111@gi.com.tr;user=phone out_uri=sip:02124440111@gi.com.tr;user=phone via_cnt==1". . # U 84.51.32.26:5060 -> 84.51.32.8:5060 ACK sip:02124440111@gi.com.tr;user=phone SIP/2.0. Via: SIP/2.0/UDP 84.51.32.26;branch=z9hG4bKaccQkWrLg. Max-Forwards: 70. From: "902247654321" ;tag=1c681629589. To: ;tag=26a82380ee921ee699cdfa26683b3165.bcf6. Call-ID: 395527579dNhC(a)84.51.32.26. CSeq: 1 ACK. Contact: . Supported: em,timer,replaces,path. Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE. User-Agent: Audiocodes-Sip-Gateway-MP-104 FXS/v.4.40.200.371. Content-Length: 0. ################################# Tried another value for from_pattern... 3(69290) val2str(): converting 84.51.32.26, 11 3(69290) PG[217] str2valp got string udp 3(69290) PG[217] str2valp got string "90224765321" select * from trusted; src_ip | proto | from_pattern -------------+-------+--------------- 84.51.32.25 | udp | "90224765321" 84.51.32.26 | udp | "90224765321" (2 rows) .

2 3

[Users] t_relay and acc logging
by Script Head 02 Mar '06

02 Mar '06

Hello, I am struggling to make sense of acc logging when combined with forwarding a call. My openser.cfg is below this in message. Every time I place a call from a softphone that registeres to SER, I get 30+ records in acc, most of them INVITEs and some are ACKs and there are no BYEs. I can't seem to find any documentation that would explain to me what I am doing wrong. Someone, please put me on the right path. route{ setflag(3); if(method=="REGISTER") { if (!www_authorize("domain.com", "subscriber")) { www_challenge("richmedium.com", "0"); return; }; save("location"); return; } if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); return; }; if(method=="INVITE") { if (!www_authorize("richmedium.com", "subscriber")) { www_challenge("richmedium.com", "0"); return; }; } if (uri=~"^sip:1[0-9]*{10}@192.168.1.10") { forward(192.168.1.10,5061); } else { sl_send_reply("404", "Use prefix 1 to dial"); } t_relay(); }

2 1

[Users] UDP -> TCP BYE goes astray
by Dr. Rodney G. McDuff 02 Mar '06

02 Mar '06

I have an * server at XXX.XXX.2.46 and an opener 1.0.1 server at XXX.XXX.2.40 and a Windows messenger 5.1 UA at YYY.YYY.152.57. When WM5 is configured to use UDP a BYE transmitted my the * server is properly transmitted to the WM5 UA via the Openser proxy. All is good. > U 2006/03/02 15:17:27.332828 XXX.XXX.2.46:5060 -> XXX.XXX.2.40:5060 > BYE sip:YYY.YYY.152.57:16478 SIP/2.0. > Via: SIP/2.0/UDP XXX.XXX.2.46:5060;branch=z9hG4bK6a5f9584;rport. > Route: <sip:XXX.XXX.2.40;ftag=8400e0fa7cbe4cb5a23e935b7d64c1f3;lr=on>. > From: <sip:55555@domain.net>;tag=as25ec0bbf. > To: "user(a)domain.net" > <sip:user@domain.net>;tag=8400e0fa7cbe4cb5a23e935b7d64c1f3;epid=1339bc4029. > Contact: <sip:55555@XXX.XXX.2.46>. > Call-ID: 45ddd129b23b48e791d5f82093a130a1. > CSeq: 102 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > > > U 2006/03/02 15:17:27.344137 XXX.XXX.2.40:5060 -> YYY.YYY.152.57:16478 > BYE sip:YYY.YYY.152.57:16478 SIP/2.0. > Record-Route: <sip:XXX.XXX.2.40;ftag=as25ec0bbf;lr=on>. > Via: SIP/2.0/UDP XXX.XXX.2.40;branch=z9hG4bK9ffb.cfbffdb2.0. > Via: SIP/2.0/UDP XXX.XXX.2.46:5060;branch=z9hG4bK6a5f9584;rport=5060. > From: <sip:55555@domain.net>;tag=as25ec0bbf. > To: "user(a)domain.net" > <sip:user@domain.net>;tag=8400e0fa7cbe4cb5a23e935b7d64c1f3;epid=1339bc4029. > Contact: <sip:55555@XXX.XXX.2.46>. > Call-ID: 45ddd129b23b48e791d5f82093a130a1. > CSeq: 102 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 69. > Content-Length: 0. > However when the WM5 UA is configured to use TCP the RURI over written by the first Route header and the BYE goes nowhere with a 404 error. Why is this so and how can it be overcome. > > U 2006/03/02 15:07:59.622462 XXX.XXX.2.46:5060 -> XXX.XXX.2.40:5060 > BYE sip:user@domain.net:8923 SIP/2.0. > Via: SIP/2.0/UDP XXX.XXX.2.46:5060;branch=z9hG4bK3fb57de2;rport. > Route: > <sip:XXX.XXX.2.40;transport=tcp;ftag=3cc4119df7fb42e9a4bccbb508466931;lr=on>. > From: <sip:55555@domain.net>;tag=as3338df1d. > To: "user(a)domain.net" > <sip:user@domain.net>;tag=3cc4119df7fb42e9a4bccbb508466931;epid=f272a56bbd. > Contact: <sip:55555@XXX.XXX.2.46>. > Call-ID: 3ff00aceaba0435a9ed23dc01dc7a77a. > CSeq: 102 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 70. > Content-Length: 0. > > > > T 2006/03/02 15:07:59.643864 XXX.XXX.2.40:42664 -> XXX.XXX.2.40:5060 [AP] > BYE > sip:XXX.XXX.2.40;transport=tcp;ftag=3cc4119df7fb42e9a4bccbb508466931;lr=on > SIP/2.0. > Record-Route: <sip:XXX.XXX.2.40;ftag=as3338df1d;lr=on>. > Via: SIP/2.0/TCP XXX.XXX.2.40;branch=z9hG4bK5419.1af2f513.0. > Via: SIP/2.0/UDP XXX.XXX.2.46:5060;branch=z9hG4bK3fb57de2;rport=5060. > From: <sip:55555@domain.net>;tag=as3338df1d. > To: "user(a)domain.net" > <sip:user@domain.net>;tag=3cc4119df7fb42e9a4bccbb508466931;epid=f272a56bbd. > Contact: <sip:55555@XXX.XXX.2.46>. > Call-ID: 3ff00aceaba0435a9ed23dc01dc7a77a. > CSeq: 102 BYE. > User-Agent: Asterisk PBX. > Max-Forwards: 69. > Content-Length: 0. > T 2006/03/02 15:07:59.656229 XXX.XXX.2.40:5060 -> XXX.XXX.2.40:42664 [AP] > SIP/2.0 404 User Not Found. > Via: SIP/2.0/TCP XXX.XXX.2.40;branch=z9hG4bK5419.1af2f513.0. > Via: SIP/2.0/UDP XXX.XXX.2.46:5060;branch=z9hG4bK3fb57de2;rport=5060. > From: <sip:55555@domain.net>;tag=as3338df1d. > To: "user(a)domain.net" > <sip:user@domain.net>;tag=3cc4119df7fb42e9a4bccbb508466931;epid=f272a56bbd. > Call-ID: 3ff00aceaba0435a9ed23dc01dc7a77a. > CSeq: 102 BYE. > Server: OpenSer (1.0.1-tls (i386/linux)). > Content-Length: 0.

2 1

[Users] how to route a call between servers
by vikram sarathy 02 Mar '06

02 Mar '06

hi, i have 3 servers in my setup, in which each server has a user registered. i need to route a call from one server to other. in first case i kept all servers in same domine and later i need to check with all servers in different domains. can any one tell me is there any way to achive this. with regards vicky vicky

2 1

[Users] NEW Feature: uac_auth() via AVPs
by Bogdan-Andrei Iancu 02 Mar '06

02 Mar '06

Hi, based on the original feature request from Thomas Gelf <thomas(a)gelf.net> (see http://www.openser.org/pipermail/devel/2006-January/001701.html), the UAC module from OpenSER head allows dynamic definition of credentials to be used in UAC authentication via AVPs. Three new module parameters allow the definitions of the AVPs to be used as realm, username and password. See http://openser.org/docs/modules/1.1.x/uac.html How it works? if the AVPs are defined (note you have to define none or all three of them), when UAC auth is performed, the module will first look for the AVPs to see if they are set or not. If all set and the realm from AVP matches the realm from challenge, the credentials from AVPs will be use. Otherwise the statically defined credentials (via module parameter) will be tried. Important Note: only the first found AVPs will be used! regards, bogdan

1 0

[Serusers] Noisy feedback tells
by Piergiorgio Venuti 02 Mar '06

02 Mar '06

Hi all, I've this problem when I start my soft phone client with ser: Warning: 392 192.168.87.252:5060 "Noisy feedback tells: pid=424 req_src_ip=192.168.87.71 req_src_port=6756 in_uri=sip:ser.mydomain.pri out_uri=sip:ser.mydomain.pri via_cnt==1" I look that when my client try to register to ser, it received an 200 ok response, but my client resend a register request again and again, this loop cause a latence in my traffic. Any idea? In attach is my config file Tnx -- +----------------------------------------------------------------------+ | Ing. Piergiorgio Venuti, CCSP Email: piergiorgio(a)mediaservice.net | | System Manager Tel: +39-011-32.72.100 | | Fax: +39-011-32.46.497 | | @ Mediaservice.net S.R.L. http://mediaservice.net | | Via S.Bernardino 17 Torino | | Disclaimer: http://@Mediaservice.net/disclaimer | +----------------------------------------------------------------------+ # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) listen=192.168.87.252 # hostname matching an alias will satisfy the condition uri==myself". alias=ser.mydomain.pri alias=ser.mydomain.net /* Uncomment these lines to enter debugging mode log_stderror=yes */ check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=7 # fifo - FIFO special file pathname fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@127.0.0.1/ser" # syn_branch - Shall the server use stateful synonym branches? It is # faster but not reboot-safe. Default is yes. syn_branch=yes # memlog - Debugging level for final memory statistics report. Default # is L_DBG -- memory statistics are dumped only if debug is set high. memlog=3 # sip_warning - Should replies include extensive warnings? By default # yes, it is good for trouble-shooting. sip_warning=yes # reply_to_via - A hint to reply modules whether they should send reply # to IP advertised in Via. Turned off by default, which means that # replies are sent to IP address from which requests came. reply_to_via=no # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/opt/ser/lib/ser/modules/sl.so" loadmodule "/opt/ser/lib/ser/modules/tm.so" loadmodule "/opt/ser/lib/ser/modules/mysql.so" loadmodule "/opt/ser/lib/ser/modules/domain.so" loadmodule "/opt/ser/lib/ser/modules/uri_db.so" loadmodule "/opt/ser/lib/ser/modules/rr.so" loadmodule "/opt/ser/lib/ser/modules/maxfwd.so" loadmodule "/opt/ser/lib/ser/modules/usrloc.so" loadmodule "/opt/ser/lib/ser/modules/registrar.so" loadmodule "/opt/ser/lib/ser/modules/textops.so" loadmodule "/opt/ser/lib/ser/modules/mediaproxy.so" loadmodule "/opt/ser/lib/ser/modules/nathelper.so" loadmodule "/opt/ser/lib/ser/modules/pa.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/opt/ser/lib/ser/modules/auth.so" loadmodule "/opt/ser/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # param settings modparam("pa", "use_db",1) modparam("pa","db_url", "mysql://ser:heslo@127.0.0.1/ser") modparam("pa", "default_expires", 3) modparam("pa","pa_domain", "ser.mydomain.pri") #modparam("domain", "db_mode", 1) modparam("domain", "db_url", "mysql://ser:heslo@127.0.0.1/ser") #modparam("domain", "domain_table", "new_name") #modparam("domain", "domain_col", "domain_col") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy", "natping_interval", 40) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "sip_asymmetrics", "/etc/ser/sip-clients") modparam("mediaproxy", "rtp_asymmetrics", "/etc/ser/rtp-clients") modparam("registrar", "nat_flag", 6) # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # 2 enables write-back to persistent mysql storage for speed # disable=0, write-through=1 # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # database location modparam("usrloc", "db_url", "mysql://ser:heslo@127.0.0.1/ser") # minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) # -- auth params -- # Uncomment if you are using auth module # modparam("auth", "secret", "johndoessecretphrase") modparam("auth", "nonce_expire", 600) modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "db_url", "mysql://ser:heslo@127.0.0.1/ser") modparam("auth_db", "user_column", "user") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); setflag(6); use_media_proxy(); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method=="BYE" || method=="CANCEL" ) { end_media_session(); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("ser.mydomain.net", "subscriber")) { # www_challenge("ser.mydomain.net", "0"); # break; # }; save("location"); # pa_handle_registration("ser.mydomain.pri"); break; }; if (uri==myself) { if (method=="SUBSCRIBE") { if (!t_newtran()) { sl_reply_error(); break; }; handle_subscription("registrar"); break; } } lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); break; } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; if (!t_relay()) { sl_reply_error(); }; }

1 0

[Users] Loadbalancing using Path-HF with NAT-Support
by Andreas Granig 02 Mar '06

02 Mar '06

Hi, I've uploaded a new module "path" to the tracker on sourceforge, which allows a intermediate proxy (an openser instance working as loadbalancer) to add it's URI into a Path-HF in various ways: - add_path(): Adds URI including the address of the outgoing interface as Path-HF in the form of "Path: <sip:1.2.3.4:5060>" - add_path_received(): like the above, but adds a parameter "received=sip:2.3.4.5:1234" to the URI to pass the received-address of an UAC to a registrar - both of the above functions also allow to pass a userpart for the Path-URI, so add_path("loadbalancer") results in a Path-HF "Path: <sip:loadbalancer@1.2.3.4:5060>" - If the module parameter "use_received" is set to 1, the module hooks into rr-processsing and sets the "received"-parameter of a Route-URI as destination-uri if available. This is used in combination with add_path_received(), where subsequent requests are sent to the received-address of the initial registration (aka the URI given in the received-parameter of the Path-HF). Also included is a patch to the registrar module, which allows to store the URI in the received-parameter of the first Path-HF as received-value of usrloc and set the NAT flag for this contact. This feature can be enabled by setting the parameter "path_use_received" of the registrar module to 1. One should be aware that nat-pinging of such a contact isn't supported by nathelper/mediaproxy yet, but the values are in place (since I use an external application for nat-pinging). Some feedback is very much appreciated. Cheers, Andy

3 5

[Serusers] multiple domains
by Maciej Świniarski 02 Mar '06

02 Mar '06

Hi, i have a little problem i am using modparam("usrloc|registrar|avpops", "use_domain", 1) i have few domains ..for example xxx.com yyy.com zzz.com i also done some numbering plan and: for xxx.com have 10XX yyy.com have 20XX zzz.com have 30XX i want to users from xxx.com can "ring" to yyy.com using sip:20XX@yyy.com or just 20XX(a)xxx.com the problem is how to setup route any connection going to numbers like 20XX from any domain will go to domain yyy.com now i have done it with aliases so i put 2001(a)xxx.com --> sip:2001@yyy.com but this suck any one can tell me how to rewritedomain in uri ? using rewritehost is good for forwarding call to other sip server .. but those domains are all in one additionally there are things like sameusername(a)xxx.com sameusername(a)yyy.com and all subscribers are in one mysql table so this must be done using some route trick. Can anyone have any idea?

1 0

[Serusers] Odd problem with mediaproxy
by Chris Roberts 02 Mar '06

02 Mar '06

Heya, I'm trying to diagnose a problem that seems to occur in ser and OpenSER regarding the mediaproxy module. In some instances, the rewrite of the c= header can create duplicates, like: o=- 3350242704 3350242704 IN IP4 y.y.y.y. ... c=IN IP4 x.x.x.xx.x.x.x. ... m=audio 3505035050 RTP/AVP 3 97 98 8 0 101. Etc. I have looked at the code and I do not believe this is a mediaproxy specific problem, however I have to say I'm confused by the 'lumps'. Am I right in thinking that these are just chunks of strings in a linked list that are reconstructed later to form the message? Any ideas on this bug? Cheers, Chris. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.1.1/271 - Release Date: 28/02/2006

3 2

[Serusers] Trouble running SER & Serweb on a local server
by Varun SR 02 Mar '06

02 Mar '06

Hi, I'm trying to run SER and SERWEB on a local server. ser runs fine, but serweb always gives an error: "Bad username or password" on logging in with admin and password heslo. I am pretty sure I've set the doman and other details properly, and uncommented the lines in ser.cfg as mentioned in INSTALL. How do I check what the problem is? Thanks. --------------------------------- Yahoo! Mail Bring photos to life! New PhotoMail makes sharing a breeze.

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RE: [Users] record_route_preset()
by Douglas Garstang 02 Mar '06

02 Mar '06

Bogdan, Version 1.0.0 of OpenSER... Actually, I found that by setting 'advertised_address', OpenSER would put that value in the Via header, and that's seems to be what Asterisk is using when it sends messages back to OpenSER. I wish I knew the exact difference between the meaning of the Record-Route and Via headers..... :O I've also found that OpenSER doesn't put the 'advertised_address' in the Via of BYE messages from phones that it passes to Asterisk. It puts the first interface in there instead. Any idea why this would be? It puts the Via from 'advertised_address' in everything else.... -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Wed 3/1/2006 11:43 AM To: Douglas Garstang Cc: users(a)openser.org Subject: Re: [Users] record_route_preset() Hi Doug, I just tested and I'm not able to reproduce your problem. The correct usage is: record_route_preset("172.31.140.205"); What version of openser are you using? if devel, be sure you have the latest CVS version. anyhow, for your purposes, the correct approach will be : force_send_socket(udp:172.31.140.205); record_route(); by this you will select the desired outgoing interface. regards, bogdan Douglas Garstang wrote: >I have OpenSER installed on a system with 4 network interfaces. When OpenSER sends out an INVITE to an Asterisk system, it puts the Record-Route header in the SIP message: > >U 216.186.140.205:5060 -> 172.31.140.203:5060 >INVITE sip:9220402@pbx1-mgt0.ipt.oneeighty.com:5060;user=phone SIP/2.0. >Record-Route: <sip:216.186.140.205;ftag=C0A1B748-F1234161;lr=on>. > >However, I would like it to put a different IP address in the Record-Route header, to force the Asterisk system at the other end to send replies back on a certain interface. I tried using record_route_preset instead of record_route, like this: > >record_route_preset("172.31.140.205"); > >Now, when SIP messages go out, they look like this: > >U 216.187.140.205:5060 -> 172.31.140.203:5060 >INVITE sip:9220402@pbx1-mgt0.ipt.oneeighty.com:5060;user=phone SIP/2.0. >Record-Route: <sip:9220402@sip:172.31.140.205:5060;ftag=D26AD271-F90F8EC6;lr=on>. > >Huh? Why has OpenSER put the to URI in the record route? That doesn't look right to me. > >I also tried using record_route_preset like this: >record_route_preset("172.31.140.205:5060"); >record_route_preset("sip:172.31.140.205"); >record_route_preset("sip:172.31.140.205:5060"); > >and it always puts the to uri in there. Why???? Would GREATLY appreciate some advice on this. >Doug. > >-----Original Message----- >From: Helge Waastad [mailto:helge@smartnet.no] >Sent: Wednesday, March 01, 2006 8:53 AM >To: Andreas.granig(a)inode.info >Cc: devel(a)openser.org >Subject: [Devel] Re: Loadbalancing using Path-HF with NAT-Support > > >Hi, >Looking forward to try it out. > >Until now, the path-header (including nat-ping script) works as a charm >for me, but this will probably make my loadbalancer much cleaner :-) > >br hw > > > > >

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[Users] * sip response 407 when using dispatcher (Help Please)
by Chan, Ka Lun 02 Mar '06

02 Mar '06

Hi All, I am trying to set up SER with Dispatcher to loadbalancing the traffic to 2 * boxes. SER was able to select the * IPs from the dispatcher.list, but * SIP response back authentication required. It work perfectly if I i use rewritehostport instead of using the dispatch module. I am pulling my hair now and still don't know where the problem at. openser.cfg if (uri=~"sip:\+?[1-9][0-9]*@.*") { ds_select_dst("2", "0"); route(4); route(5); return; }; route[4] { if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } route[5] { setflag(1); t_on_reply("1"); forward(uri:host, uri:port); append_hf("P-hint: main PSTN route\r\n"); t_on_failure("1"); if (!t_relay()) { sl_reply_error(); return; }; } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } failure_route[1] { append_hf("P-hint: backup PSTN route\r\n"); rewritehost("x.x.x.x"); rewriteport( "5060"); append_branch(); t_relay(); } SIP.conf form * [general] host=dynamic bindaddr=0.0.0.0 port=5060 useragent=x context=default disallow=all allow=g729 allow=ulaw autocreatepeer=yes dtmfmode=rfc2833 qualify=no nat=yes canreinvite=no Retransmitting #5 (no NAT): SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP x.x.x.x;branch=0 Via: SIP/2.0/UDP 192.168.1.107:5060;received=x.x.x.x;branch=z9hG4bK263724 From: "testing" <sip:testing@64.127.123.100>;tag=5318 To: <sip:exten@x.x.x.x>;tag=as1cf1692c Call-ID: 1141227578-724-TF-GIXXER(a)192.168.1.107 CSeq: 813 INVITE User-Agent: x Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:exten@x.x.x.x> Proxy-Authenticate: Digest realm="asterisk", nonce="40a9764f" Content-Length: 0

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[Serusers] classkit.so module php5
by Ofer Zaarur 01 Mar '06

01 Mar '06

Hi All, I have problem getting classkit module to set serweb correctly using php5. it looks like this issue had been raise before. please let me know where I can get it. thanks, O. enclose is the email that discuss this issue: ---------------------------------------------------------------------------------------------- That't the package that I created, it is the same as the one in the repository. Thanks. Jan. On 11-07-2005 21:45, Gareth C. Fowler wrote: >* I did install from an RPM file, but I didn't get it from your repository. *>* For the life of me, I can't remember where I downloaded it from, but I have *>* included it on as an attachment. *>* *>* G. *>* *>* -----Original Message----- *>* From: Jan Janak [mailto:jan at iptel.org<http://lists.iptel.org/mailman/listinfo/serusers> ] *>* Sent: 11 July 2005 21:38 *>* To: Gareth C. Fowler *>* Cc: serusers at iptel.org<http://lists.iptel.org/mailman/listinfo/serusers> *>* Subject: Re: RE: [Serusers] Try install Classkit extension *>* *>* *>* Out of curiosity, could you let me know which package does provide the *>* extension ? (run rpm -qf /usr/lib/php/modules/classkit.so). Is it the *>* one that is available from our yum repository or did you install it some *>* other way ? *>* *>* I am asking because I could not find classkit extension packaged for FC4 *>* so I had to create the package, but if it exists somewhere else then I *>* would drop the package. *>* *>* Jan. *>* *>* On 11-07-2005 21:03, Gareth C. Fowler wrote: *

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[Users] acc module and logging incomplete calls
by Script Head 01 Mar '06

01 Mar '06

My SER distributes calls to a bunch of Asterisk servers. Here's my route: route{ setflag(1); if (!mf_process_maxfwd_header("10")) { log("LOG: Too many hops\n"); sl_send_reply("483","Too Many Hops"); return; }; if (loose_route()) { t_relay(); return; }; if(uri==myself) { if(method=="REGISTER") { save("location"); return; }; }; if(method=="INVITE") { record_route(); if (uri=~"^sip:1[0-9]*{10}@192.168.1.10") { ds_select_dst("2", "0"); forward(uri:host, uri:port); return; }; }; } I am trying to log all calls including failed, busy, congestion, incomplete using acc. Strangely, in my acc table I only see records affiliated with BYE and not with INVITE. Thru, all I see are successful calls. Is there a way I can make acc log incomplete calls?

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[Users] record_route_preset()
by Douglas Garstang 01 Mar '06

01 Mar '06

I have OpenSER installed on a system with 4 network interfaces. When OpenSER sends out an INVITE to an Asterisk system, it puts the Record-Route header in the SIP message: U 216.186.140.205:5060 -> 172.31.140.203:5060 INVITE sip:9220402@pbx1-mgt0.ipt.oneeighty.com:5060;user=phone SIP/2.0. Record-Route: <sip:216.186.140.205;ftag=C0A1B748-F1234161;lr=on>. However, I would like it to put a different IP address in the Record-Route header, to force the Asterisk system at the other end to send replies back on a certain interface. I tried using record_route_preset instead of record_route, like this: record_route_preset("172.31.140.205"); Now, when SIP messages go out, they look like this: U 216.187.140.205:5060 -> 172.31.140.203:5060 INVITE sip:9220402@pbx1-mgt0.ipt.oneeighty.com:5060;user=phone SIP/2.0. Record-Route: <sip:9220402@sip:172.31.140.205:5060;ftag=D26AD271-F90F8EC6;lr=on>. Huh? Why has OpenSER put the to URI in the record route? That doesn't look right to me. I also tried using record_route_preset like this: record_route_preset("172.31.140.205:5060"); record_route_preset("sip:172.31.140.205"); record_route_preset("sip:172.31.140.205:5060"); and it always puts the to uri in there. Why???? Would GREATLY appreciate some advice on this. Doug. -----Original Message----- From: Helge Waastad [mailto:helge@smartnet.no] Sent: Wednesday, March 01, 2006 8:53 AM To: Andreas.granig(a)inode.info Cc: devel(a)openser.org Subject: [Devel] Re: Loadbalancing using Path-HF with NAT-Support Hi, Looking forward to try it out. Until now, the path-header (including nat-ping script) works as a charm for me, but this will probably make my loadbalancer much cleaner :-) br hw -- Helge Waastad Senior Konsulent Systemavdelingen Smartnet _______________________________________________ Devel mailing list Devel(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/devel

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[Users] OT - Cisco IP Phone and PC in diferent VLANs (with 802.1x)
by Joao Pereira 01 Mar '06

01 Mar '06

Hello to all I would like to know If some of you have already configured an Cisco IP Phone (7940 or 7960) to work in a different VLAN than the PC that is connected through the phone switch? I know that this can be done with the Skinny firmware, but I dont if it works with the SIP firmware. The Cisco technical staff told me that these phones dont support 802.1x but can work as pass-through. This way I can still use the PCs with 802.1x and the phones in the same Ethernet plug. Did someone made it with the Cisco IP phones? What configuration do I need in the phones and in the switch? Thanks Joao Pereira

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[Users] NEW Functionality: pseudo-variables in authentication modules
by Bogdan-Andrei Iancu 01 Mar '06

01 Mar '06

Hi, the devel branch support the usage of pseudo variables for the "realm" parameter of all auth functions - www/proxy-challenge/authorize. This allow dynamic injection of realm name from outside the SIP message. This feature was originally requested by Martin Koenig <martin.koenig(a)toplink.de>. For details, see: http://www.openser.org/pipermail/devel/2006-January/001815.html CVS and online docs are updated. regards, bogdan

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[Users] MySQL5.0.18 + OpenSER1.0.1 - Looking on HowTo...
by Edson 01 Mar '06

01 Mar '06

In a previous e-mail I described that I only had success on compiling OpenSER 1.0.0-tls against MySQL 4.1.13. Not against MySQL 5.0.18. With the release of 1.0.1, I return to give a try, but didn't have success. Looking at Google, I find the kind of problem on SER-DEV discussion list (http://mail.iptel.org/pipermail/serdev/2005-April/004343.html), but it didn't work for me, so I move looking. Than I find a bug-report on MySQL (number 13159 - http://bugs.mysql.com/bug.php?id=13159) and try they last advise: include '-fPIC' on /usr/lib/rpm/rpmrc and rebuild MySQL RPMs. After that the compilation finished well, as the installation. The problem now is on module load. I got this message complaining on a 'floor' reference: 0(19939) ERROR: load_module: could not open module </usr/lib/openser/modules/mysql.so>: /usr/lib/openser/modules/mysql.so: undefined symbol: floor I'm running out of ideas.. :'-( Has anybody got to install OpenSER on a Xeon machine with MySQL 5.0.18 support? Any help would be appreciated. Edson.

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[Users] pseudo-variables notation update
by Daniel-Constantin Mierla 01 Mar '06

01 Mar '06

Hello, in order to allow names with different length, the pseudo-variables notation is similar with the one in make. You can enclose the name in between '(' and ')', e.g. $ru is the same with $(ru). This comes with the support of registering new pseudo-variables dynamically from modules. As an example, tm module exports $T_branch_idx which refers to the branch index for which is executed the branch_route[]. In case when parenthesis are not used, the name of the pseudo variable is considered to end at the first character which is not digit, letter, "." or "_". Also, the notation for avp names has changed to avoid confusions (see:http://openser.org/pipermail/devel/2006-February/001943.html ). Now the following notation must be used: - $avp(s:avp_name) - refers to the avp having the string name 'avp_string' - $avp(i:avp_id) - refers to the avp having the integer id 'avp_id' - $avp(avp_alias) - refers to the avp defined by the alias 'avp_alias' Cheers, Daniel

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[Serusers] SER Manual
by Marcio Sousa 01 Mar '06

01 Mar '06

Hi, Where can I find some manual for SER wich can explain how to work with SER srcipts? Thnks.

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RE: [Serusers] Saving historical registration information
by Vitaly Nikolaev 01 Mar '06

01 Mar '06

Or to use exec command that will get REGISTER as stdin, parse it and save... if you do not have thousands of customers which register every 40 sec then it should work though it not a nice solution, I agree, just easiest to do :) using avpops you can only do that for customer who really wants it. -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Dr Rodney G. McDuff Sent: Wednesday, March 01, 2006 6:40 AM To: serusers Subject: Re: [Serusers] Saving historical registration information Jan Janak wrote: > Unfortunately this is not possible without usrloc module modifications. > Use a database backend that has triggers. When a new entry is added to the location table write the record somewhere else. > Jan. > > Alistair Cunningham wrote: > >> We've been asked by a customer to keep the last registration >> information, even after expiry. >> >> They want to be able to look at the database and see where and when a >> phone last registered from, even if it was months ago. Only the last >> registration is important; they don't care about older ones. >> >> This could be done using various methods such as a trigger in the >> database to update another table any time location is written to, but >> this is ugly. >> >> Does any one know of a simple way to do it, preferably in ser.cfg? >> >> > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > -- Dr. Rodney G. McDuff |Ex ignorantia ad sapientiam Manager, Strategic Technologies Group| Ex luce ad tenebras Information Technology Services | The University of Queensland | EMAIL: mcduff(a)its.uq.edu.au | TELEPHONE: +61 7 3365 8220 | _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] CFP: Third VoIP Security Workshop, Deadline extended
by Sven Ehlert 01 Mar '06

01 Mar '06

Third Annual VoIP Security Workshop In Cooperation with ACM SIGCOMM Berlin, Germany 01, 02 June 2006 www.iptel.org/voipsecurity Submission Deadline: 15.03.06 This security workshop is the third following successful events in Dallas 2004 and Washington 2005. The workshop addresses the challenges, advanced technologies and regulations for providing secure VoIP services in the Internet as well as in third generation networks. The 2006 workshop will extend the scope of VoIP security to IMS services and present the latest advances in denial of service and SPIT detection and protection mechanisms. The format of the workshop is based on two-day single-track sessions, with presentations of invited and regular papers from academia and industry. The papers solicited in the Third VoIP Security workshop cover a variety of topics including but not limited to: * Intrusion and anomaly detection of VoIP and IMS systems * SPIT/SPIM detection and prevention * Threat and vulnerability analyses of VoIP and IMS systems * Regulatory aspects of VoIP and IMS communication * Securing Peer-to-peer systems * Securing service creation environments * Privacy and anonymity * Monitoring and testing tools for VoIP and IMS infrastructures * Security of VoIP supporting systems such as AAA and ENUM * Support of lawful interception and securing emergency services in VoIP and IMS services * High availability solutions for VoIP and IMS systems The Third VoIP Security workshop is a two day workshop with a special session of invited papers from industry and government. The program committee will referee all papers, and accepted papers will be published in the conference proceedings Important Dates: ======================= * Full paper submission deadline: March 15, 2006 * Notification of acceptance: April 10, 2006 * Camera ready submission: May 01, 2006 General Chair: ======================= Radu Popescu-Zeletin Fraunhofer Fokus Program Chairs: ======================= Dorgham Sisalem Tekelec Ram Dantu University of North Texas Henning Schulzrinne Columbia University Local Organization Chair: ======================= Sven Ehlert Fraunhofer Fokus Technical Program Committee ======================= Ralf Ackermann TU Darmstadt Hossam Afifi INT Evry Elisa Bertino Purdue University George Carle Tübingen University Jasson Casey Alcatel Ram Dantu University of North Texas Sven Ehlert Fraunhofer Fokus Sonia Fahmy Purdue University Dipak Ghosal University of California Davis Günter Müller University of Freiburg Saverio Niccolini NEC Labs Europe Jörg Ott Helsinki University of Technology Ozgur Gurleyen Vodafone Basavaraj Patil Nokia Neeli Rashmi Prasad University of Aalborg Pradeep Samudra Samsung Peter Schoo DoCoMo Euro-Labs Henning Schulzrinne Columbia University Dorgham Sisalem Tekelec Jeff Smith DHS Burkhard Stiller University of Zurich and ETH Zurich Duminda Wijesekeara George Mason University

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[Serusers] How to remove Contact HF in 200 reply for REGISTER
by Arek Bekiersz 01 Mar '06

01 Mar '06

Hi, How to disable sending Contact HF in OK send to a successful (i.e. authenticated) REGISTER request? Just use remove_hf() from textops or there is other option? I've got stupid UA that behaves badly when working behind NAT. First it sends REGISTER without anything (no Contact, Expires). On successful authentication SER responds 200 OK with Contact HF, set to public IP, exactly how SER sees UA behind NAT (like some_IP:61001). Then the stupid UA sends another REGISTER message, this time with Contact and Via HF set to this public value (i.e. some_IP:61001). It spoils my nathelper/rtpproxy detection of devices behind NAT. Of course I can enable it by regexp on User-Agent HF, but his is not an option, as plenty of those UAs runs not behind NAT. -- Regards, Arek Bekiersz

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[Users] openser & sems memory error
by Ibrahim TUNALI 01 Mar '06

01 Mar '06

Hi, i installed openser (1.0 stable) and sems (0.9 stable). I recive "ERROR:tm:write_to_fifo: writev failed: Cannot allocate memory" error when openser send request to sems voice mail. Is it compile error or do you have any solution for this. regards openser voicemail seciton; -------------------------- # switch to stateful mode: if (!t_newtran()){ sl_send_reply("500","could not create transaction"); return; }; # prevent timeout on the other side: t_reply("100","Trying - just wait a minute !"); # actively absorb ACKs if (method == "ACK") { t_relay(); return; } if (method=="INVITE"){ # no service number, redirect to voicemail. # do not forget to load AVPs so that voicemail gets the # callee's email address. avp_db_load("$ruri","i:34/$email_scheme"); if(!t_write_req("/tmp/am_fifo","voicemail/vm_email")) { t_reply("500","error contacting sems"); }; return; } else if (method=="BYE" || method=="CANCEL") { # Sems should already know which plug-in is handling that # call. "bye" is no plug-in name. It is a reserved name which # tells Sems to terminate the call. if(!t_write_req("/tmp/am_fifo","bye")) { t_reply("500","error contacting sems"); }; }; openser log; ------------ Feb 23 17:18:17 voip /usr/sbin/openser[28398]: ERROR:tm:write_to_fifo: writev failed: Cannot allocate memory Feb 23 17:18:17 voip /usr/sbin/openser[28398]: ERROR:tm:t_write_req: write_to_fifo failed Feb 23 17:18:17 voip /usr/sbin/openser[28396]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:18 voip /usr/sbin/openser[28392]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:20 voip /usr/sbin/openser[28396]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:24 voip /usr/sbin/openser[28398]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:28 voip /usr/sbin/openser[28394]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:32 voip /usr/sbin/openser[28398]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:36 voip /usr/sbin/openser[28394]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:40 voip /usr/sbin/openser[28398]: Warning: sl_send_reply: I won't send a reply for ACK!! Feb 23 17:18:44 voip /usr/sbin/openser[28394]: Warning: sl_send_reply: I won't send a reply for ACK!! sems log; ------------ Feb 23 17:19:45 voip Sems[28477]: Error: (AmUtils.cpp)(str2i)(147): str2i: unexpected char s in ser Feb 23 17:19:45 voip Sems[28477]: Error: (AmUtils.cpp)(str2i)(147): str2i: unexpected char s in ser Feb 23 17:19:45 voip Sems[28480]: Debug: (sems.cpp)(main)(386): Loading application plug-ins Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/apps/announcement.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAppPlugIn)(281): application 'announcement' loaded. Feb 23 17:19:45 voip Sems[28480]: Debug: (SemsConfiguration.cpp)(reloadModuleConfig)(70): using inline configuration for module announcement Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/apps/bot.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAppPlugIn)(281): application 'bot' loaded. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmServer.cpp)(registerFct)(401): AmUnServer::registerFct: function 'bot_start' has been registered. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmServer.cpp)(registerFct)(208): AmFifoServer::registerFct: function 'bot_start' has been registered. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/apps/conference.so ... Feb 23 17:19:45 voip Sems[28482]: Debug: (AmThread.cpp)(_start)(63): Thread 28482 is starting. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmThread.cpp)(start)(85): Thread 28482 is just created. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAppPlugIn)(281): application 'conference' loaded. Feb 23 17:19:45 voip Sems[28480]: Debug: (SemsConfiguration.cpp)(reloadModuleConfig)(70): using inline configuration for module conference Feb 23 17:19:45 voip Sems[28480]: Warning: (Conference.cpp)(onLoad)(77): no join_sound specified in configuration Feb 23 17:19:45 voip Sems[28480]: Warning: (Conference.cpp)(onLoad)(78): file for module conference. Feb 23 17:19:45 voip Sems[28480]: Warning: (Conference.cpp)(onLoad)(84): no drop_sound specified in configuration Feb 23 17:19:45 voip Sems[28480]: Warning: (Conference.cpp)(onLoad)(85): file for module conference. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/apps/echo.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAppPlugIn)(281): application 'echo' loaded. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/apps/isdngw.so ... Feb 23 17:19:45 voip Sems[28480]: Info: (IsdnConfiguration.cpp)(reload)(91): isdngw version 0.2 loading (supported backends: isdn4linux) ... Feb 23 17:19:45 voip Sems[28480]: Warning: (SemsConfiguration.cpp)(reloadModuleConfig)(66): no configuration found for module isdngw, maybe you want to specify config.isdngw in config file. Feb 23 17:19:45 voip Sems[28480]: Info: (IsdnGatewayFactory.cpp)(IsdnGatewayFactory)(81): isdngw: isdn4linux support disabled in configuration Feb 23 17:19:45 voip Sems[28480]: Info: (IsdnGatewayFactory.cpp)(IsdnGatewayFactory)(148): isdngw initialization successful. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAppPlugIn)(281): application 'isdngw' loaded. Feb 23 17:19:45 voip Sems[28480]: Debug: (IsdnGatewayFactory.cpp)(onLoad)(199): registering fifo function isdninfo for web access... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmServer.cpp)(registerFct)(208): AmFifoServer::registerFct: function 'isdninfo' has been registered. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/apps/number_reader.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAppPlugIn)(281): application 'number_reader' loaded. Feb 23 17:19:45 voip Sems[28480]: Debug: (SemsConfiguration.cpp)(reloadModuleConfig)(70): using inline configuration for module number_reader Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/apps/voicemail.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAppPlugIn)(281): application 'voicemail' loaded. Feb 23 17:19:45 voip Sems[28480]: Debug: (SemsConfiguration.cpp)(reloadModuleConfig)(70): using inline configuration for module voicemail Feb 23 17:19:45 voip Sems[28480]: Debug: (AnswerMachine.cpp)(loadEmailTemplates)(103): loading /etc/sems//default.template ... Feb 23 17:19:45 voip Sems[28480]: Debug: (sems.cpp)(main)(390): Loading audio plug-ins Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/audio/gsm.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(218): codec id 3 inserted Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(233): payload 'GSM'inserted with id 3 Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/audio/ilbc.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(218): codec id 4 inserted Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(237): payload 'iLBC'inserted with id 96 Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(257): file format iLBC inserted Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/audio/speex.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(218): codec id 6 inserted Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(237): payload 'speex'inserted with id 96 Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(load)(87): loading /usr/lib/sems/plug-in/audio/wav.so ... Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(218): codec id 1 inserted Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(218): codec id 2 inserted Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(233): payload 'PCMU'inserted with id 0 Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(233): payload 'PCMA'inserted with id 8 Feb 23 17:19:45 voip Sems[28480]: Debug: (AmPlugIn.cpp)(loadAudioPlugIn)(257): file format Wav inserted Feb 23 17:19:45 voip Sems[28480]: Debug: (sems.cpp)(main)(394): Starting session container Feb 23 17:19:45 voip Sems[28486]: Debug: (AmThread.cpp)(_start)(63): Thread 28486 is starting. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmThread.cpp)(start)(85): Thread 28486 is just created. Feb 23 17:19:45 voip Sems[28480]: Debug: (sems.cpp)(main)(397): Starting mailer Feb 23 17:19:45 voip Sems[28480]: Debug: (AmThread.cpp)(start)(85): Thread 0 is just created. Feb 23 17:19:45 voip Sems[28480]: Debug: (sems.cpp)(main)(400): Starting RTP scheduler Feb 23 17:19:45 voip Sems[28480]: Debug: (AmThread.cpp)(start)(85): Thread 0 is just created. Feb 23 17:19:45 voip Sems[28480]: Debug: (sems.cpp)(main)(403): Starting ICMP watcher Feb 23 17:19:45 voip Sems[28480]: Debug: (AmThread.cpp)(start)(85): Thread 0 is just created. Feb 23 17:19:45 voip Sems[28480]: Debug: (AmThread.cpp)(start)(85): Thread 0 is just created. Feb 23 17:19:45 voip Sems[28480]: Info: (AmServer.cpp)(run)(147): FIFO server started Feb 23 17:19:45 voip Sems[28487]: Debug: (AmThread.cpp)(_start)(63): Thread 28487 is starting. Feb 23 17:19:45 voip Sems[28488]: Debug: (AmThread.cpp)(_start)(63): Thread 28488 is starting. Feb 23 17:19:45 voip Sems[28489]: Debug: (AmThread.cpp)(_start)(63): Thread 28489 is starting. Feb 23 17:19:45 voip Sems[28489]: Error: (AmIcmpWatcher.cpp)(run)(47): ICMP Watcher: could not create RAW socket: Operation not permitted Feb 23 17:19:45 voip Sems[28489]: Error: (AmIcmpWatcher.cpp)(run)(48): ICMP Watcher: try to run SEMS as root or suid. Feb 23 17:19:45 voip Sems[28489]: Debug: (AmThread.cpp)(_start)(68): Thread 28489 is ending. Feb 23 17:19:45 voip Sems[28490]: Debug: (AmThread.cpp)(_start)(63): Thread 28490 is starting. Feb 23 17:19:45 voip Sems[28490]: Info: (AmServer.cpp)(run)(321): Unix socket server started

2 1

[Serusers] Proposal for new features
by Arek Bekiersz 01 Mar '06

01 Mar '06

Hi, As my current work may be interesting for the rest, I wonder if you will find those new features useful. Every opinion is appreciated. Some time ago I commited a small 'ldap' module, all-in-one. As there was some (small :-) ) feedback from people using this module, I propose to split it into following modules: 1) auth_ldap: set of authentication routines, working with standard or user-defined Ldap schemas 2) group_ldap: group membership module based on Ldap backend 3) uri_ldap: uri mangling module, based on ldap backend 4) avp_ldap: (new) module for operating on AV-pairs from Ldap, working with both user-defined or standard dbase schemas Later, it is possible to write small SER 'soap' modules, using one of freely available C/C++ soap libraries. Those modules will perform various tasks: 5) soap_ipdr: module for reporting service usage information (billing) in XML, to be used by external billing systems. Information will include all call attempts (also failed ones like CC,CAD,UCN,UCI and possibly CID completion codes from IPDR's VoIP service definition). It will return following XML documents (using SOAP as transport): * IPDRDoc - usage for specific user (user@domain) * IPDRSettlementDoc - usage for domain NOTE: that using IPDRSettlementDoc for domain is based on model where our cooperants have one virtual SIP domain. it is then easiest way to make settlement using IPDRSettlementDoc for one domain. 6) soap_parlayx: Such module would act as a gateway between SER and external Parlay X Web Services software. When called, this module will analyze SIP request and forward it to external Parlay X software. That software will fulfill request and return ReturnValue, with specific Action. Then SER soap_parlayx module will perform desired action on SIP request. More on Parlay X Web services v.2.0 can be found at: http://www.parlay.org/en/specifications/ NOTE: This behaviour mimics a hypothetic SIP-CGI gateway, but this time it is a SIP-SOAP gateway. EXAMPLE: could be the HandleBusy operation from "Part 3: Call Notification" of Parlay X Web services: 1. SIP encounters "486 Busy" situation 2. SER Script writer calls parlayx_handle(HandleBusy) function from soap_parlayx module 3. soap_parlayx module calls external software, sending handleBusyRequest, containing CallingParty, CalledParty 4. external software returns handleBusyResponse with Action structure, containing ActionToPerform (like 'route' on not route), RoutingAddress, Charging (charge for using of service) 5. SER soap_parlayx module modifies SIP request and forwards it to voicemail_SIP location. In addition it stores relevant info in IPDR data recorder for later use. External Parlay X software will be basically a SOAP engine, it can be written in any language. Unlike a soap_parlayx module itself, that engine will not be a part of SER. 7) SIP SOAP initiative Seems that IP world is slowly evolving in direction of service enabled architecture, usually based on SOAP/UDDI standards. For this reason maybe it is worth to initiate work on SIP SOAP draft standard? It would basically mimic existing IETF SIP CGI draft standard, but with far greater opportunities for creating new services, that SOAP standard brings. It will beautifuly fulfil current work in progress in my company and an European project, codename VISP (Virtual Internet Service Provider), that we are coordinator of. Project is focusing on enabling smooth company-to-company interoperability in mutual creation of new services in fast changing IP world. Including VoIP. More information can be found on Web site: http://www.visp-project.org/ -- Regards, Arek Bekiersz

4 6

[Users] Using t_reply()
by Douglas Garstang 01 Mar '06

01 Mar '06

Why can't I do this??? Arrgh! I want to be able to call onreply_route[35] when we get a reply. So, not surprisingly, in onreply_route[35] I want to see what sort of reply I got, and then send ANOTHER reply back to the originater of the message telling them something was wrong. Why can't I use t_reply in a reply_route? What's the most likely away around this, because I sure must be missing something fundamental. route[10] { t_on_failure("11"); t_on_reply("35"); rewritehostport("pbx1-mgt0.ipt.oneeighty.com:5060"); if ( !t_relay() ) { xlog ("L_INFO","t_relay() returned error"); } else { xlog ("L_INFO","t_relay() returned ok"); } } onreply_route[35] { xlog ("L_INFO","Got some sort of reply"); if (t_check_status("603")) { xlog ("L_INFO","got a decline"); t_reply("603","Declined"); } return; }

2 1

[Users] Checking reply status
by Douglas Garstang 01 Mar '06

01 Mar '06

How can I check for SIP responses, say 'Decline' for example? Do I have to use t_check_status()? I tried but I think it wants me to create a transaction first, and seeing as though the OpenSER docs are so bad, I don't know how to do this. Is there a stateless way to check for a SIP response? In general, it seems as though you can't check SIP response messages. There's the 'if method = <something>' approach, say for example 'if method == INVITE', but I have not seen any examples of how you can check for SIP response messages just as easy in a stateless fashion. Say, something like 'if response == DECLINE' or 'if response == 'Not Found'. How can this be done??? Thanks, Doug.

2 1

[Users] TLS DH cipher problem
by Tóth Péter 01 Mar '06

01 Mar '06

Helo! I find a list, where the same problem occured with EDH-RSA-DES-CBC-SHA cipher... http://marc.theaimsgroup.com/?l=openssl-users&m=109844857431850&w=2 The offered solution is the following: "You need to include a call to SSL_CTX_set_tmp_dh() which the DH parameters too." How can I include the above? Thans: Tóth Péter

2 1

[Users] How about this..
by Helge Waastad 01 Mar '06

01 Mar '06

Today, just some refelctions... A colleague of mine, early today, started getting spam messages on his Skype account, and this got me thinking: What about spamfiltering on proxy level? Has anyone tried to execute a spamassassin script on, for instance, a MESSAGE and either forwarded or discarding the call? I know this would have a deep impact on processing time, but anyway, I thought this was an interesting scenario.. br hw -- Helge Waastad Senior Konsulent Systemavdelingen Smartnet

2 1

[Serusers] Saving historical registration information
by Alistair Cunningham 01 Mar '06

01 Mar '06

We've been asked by a customer to keep the last registration information, even after expiry. They want to be able to look at the database and see where and when a phone last registered from, even if it was months ago. Only the last registration is important; they don't care about older ones. This could be done using various methods such as a trigger in the database to update another table any time location is written to, but this is ugly. Does any one know of a simple way to do it, preferably in ser.cfg? -- Alistair Cunningham, Integrics Ltd, +44 20 799 39 799 sip:acunningham@integrics.com http://integrics.com/

4 3

[Serusers] SDP of 180
by Ever Zalazar 01 Mar '06

01 Mar '06

Hi people. I want to know if it's possible to configure SER to modify the SDP of the 180. Best Regards Ever Zalazar

2 1

[Users] Decision to use OpenSER - SER
by Gerry 01 Mar '06

01 Mar '06

(I hope the subject is appropriate here and I am not starting a religious war. If the subject is inappropriate please feel to delete the message admins.thanks) We are about to make the decision to use SER or OpenSER. Are there any strong reasons why one should avoid using one or the other ? My general impression is that OpenSER tries to be more aggressive in implementing new features. However the questions regarding stability and commitment by users to the fork linger of course in my mind too. I would be most grateful if some could provide me some arguments regarding the matter. TIA for your input. Gerry

2 1

[Serusers] Ser + SEMS + set time.
by tstafin＠op.pl 01 Mar '06

01 Mar '06

Hello everyone! I'd like to configured ser that for example from 7 a.m. to 15 p.m. every call to uri 000 is forward to sems (announcement) and play some wav. I try to find module to do this but i'm stuck. Can anyone help me. Sorry for my bad english. Thanks for any help. Best regards Swaper

3 2

RE: [Serusers] Ser + SEMS + set time.
by pankaj.munjal＠wipro.com 01 Mar '06

01 Mar '06

Hi Swaper, You can use the semstalkflite module of sems. you'll need to install flite (a speech synthesizer) for the same. The way this module works is as follows: You have to send a Invite to sems with the text that you want to be played back, as a Request URI Cheers, Pankaj. -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of tstafin(a)op.pl Sent: Wednesday, March 01, 2006 1:51 PM To: serusers(a)lists.iptel.org Subject: [Serusers] Ser + SEMS + set time. Hello everyone! I'd like to configured ser that for example from 7 a.m. to 15 p.m. every call to uri 000 is forward to sems (announcement) and play some wav. I try to find module to do this but i'm stuck. Can anyone help me. Sorry for my bad english. Thanks for any help. Best regards Swaper _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[Serusers] SER and Asterisk on FOSDEM Brussels, 25 February 2006
by Arek Bekiersz 01 Mar '06

01 Mar '06

Hi, I just returned from FOSDEM, held today at ULB Free University, Brussels. It was a great event, helping to encourage growth of free software. You could see many designers and developers working to make Open Source better. I must say I "recharged my batteries", raising my spirit a little. I joined SER presentation by Jan Janak, held at 14:00 at "Chavanne" room, building "U". Jan's appearance was recepted very warmly by audience. Presentation was professional and it fitted well in schedule, still leaving a lot of time for questions from the public. I must say that new SER logo (the big "S") looked very good on the big screen, during presentation. Jan presented origins of SER project, overview of architecture and functionality and also few typical SER installation scenarios. For those knowing less about SER or willing to learn it was a best primer. He also presented answers for frequent questions like "what SER is and what is not" and what can be done to get additional functionalities. There is a photograph attached to this letter. It's was taken with 3G phone but you can still see Jan on the right, with one of his slides. :-) After SER there was Asterisk presentation and workshop, held by Mark Spencer and his colleagues. He's a great developer too and in addition he revealed his showman skills, adding many jokes for the audience to laugh at. I was particularly disarmed by funny voice prompts with Asterisk's IVR voice of Alice, although one could found some them a little abrupt :-) -- Regards, Arek Bekiersz mailto:arek@perceval.net

3 5

[Serusers] Ser with Asterisk, Transfer problem
by tdeillon＠cri74.org 01 Mar '06

01 Mar '06

Hello, I have the same problem that here http://bugs.digium.com/view.php?id=3710 but with using a ser and an Asterisk. There is a solution using ser to solve this problem ? Thanks, Thomas

1 0

[Serusers] fix contact when ser is behind nat
by Jack Wei 01 Mar '06

01 Mar '06

Hi, How do you fix the UA's contact when SER and UA are behind different NATs? When I use nathelper, it changes the contact to the source ip and port of the packet coming the NAT device...not UA's public IP and port. Currently, I have 2 SER boxes behind a load-balancing switch, and would like to get UA behind another NAT working. THanks in advance. Jack __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] Credentials problem for calling out
by mkumar＠mantragroup.com 01 Mar '06

01 Mar '06

Hi, I have installed SER on a machine and I have mysql DataBase elsewhere. I was able to add new users to that DB using serctl and even chenge their group to ld. All the new users created are able to register but when I check whether they are in ld and can call out in ser.cfg it always tells that they have no permission and not in ld. I think it is checking localhost DB for credentials to call out but I am sure for registration it is checking DB on other machine. I included there in my ser.cfg modparam("usrloc", "db_url", "mysql://user:secret@othermachineip/ser") modparam("auth_db", "db_url", "mysql://user:secret@othermachineip/ser") modparam("acc", "db_url", "mysql://user:secret@othermachineip/ser") Please tell me what more should I include for making it point to DB on other machine even for checking credentials for calling out. Thanks, Manoj.

1 1

[Users] debian core dump configuration
by Klaus Darilion 01 Mar '06

01 Mar '06

Hi! I've installed using the new debian installer from Dan - looks nice. But I have one question: If core dumps are enabled, openser will runs as user root. Is this really necessary? Wouldn't it be sufficient to start as user openser, but change the core dump directory to /tmp ? regards klaus

2 1

[Serusers] help needed
by Marcio Sousa 01 Mar '06

01 Mar '06

Hi, I would like to know if there´s a way to give "inteligence" to a sip proxy (SER) of recognizing that 2 or more clients are requesting for the same service (audio or video) and redirect the client´s call to another proxy (multicast pheraps) without closing their session previosly estabilished, aplying the model offer/answer [RFC 3264] if necessary. Thanks.

1 0

[Serusers] SER ,Asterisk and MWI
by Sharon 28 Feb '06

28 Feb '06

hello, I am trying to pass MWI from Asterisk to SER.my user agents register with Ser.i am not able to figure out how to do this. i added the changes for mailbox in sip.conf for ser peer entry. [ser] type=friend .... mailbox=XYZ also changes in chan_sip.c for asterisk but not seeing the notify messages hitting my ser server. any suggestions please, i would highly appreciate. Thanks, AA

2 1

[Users] Doesn't create .pid file and a bunch of <defunct> processes.
by Script Head 28 Feb '06

28 Feb '06

Hello, I have a generic setup of openser 1.0.1 compiled with the PostgreSQL module on Slackware 10.2 and looks like all the modules load right. When I run it in the no-fork debug mode it works fine. When I set it to fork, it doesn't write the /var/run/openser.pid file and I have a 8 [openser] <defunct> processes running. I start it like so ./openser -f /etc/openser/openser.conf -P /var/run/openser.pid Where should I start looking? ScriptHead

2 2

[Users] APT repository for Debian
by Bogdan-Andrei Iancu 28 Feb '06

28 Feb '06

Hi, starting with 1.0.1 version, Debian packages can be directly fetch via APT from the project repository. The packages are available for sarge (stable), etch (testing) and sid (unstable). The repository includes also all related radius packages: deb http:/www.openser.org/debian sarge main deb http:/www.openser.org/debian etch main deb http:/www.openser.org/debian sid main all packages are available for i386 architecture. For amd64 only unstable version is available. regards, bogdan

1 0

Re: [Serusers] "Getting Started" configs
by Andrey Kouprianov 28 Feb '06

28 Feb '06

Says that pid file was not found and ser could not start (usual serctl message if ser fails to start).. Btw, Im using ser_0.9.3. I understand that I can start SER w/o serctl, but I am just curious why is it happening... Eagerly waiting for your reply :)) Andrey. On 2/27/06, Christian Thomas <cthomas(a)canalwest.com> wrote: > Hi, > > What error did you get ? > > Christian > > -----Original Message----- > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]On > Behalf Of Andrey Kouprianov > Sent: segunda-feira, 27 de fevereiro de 2006 10:17 > To: serusers(a)lists.iptel.org > Subject: [Serusers] "Getting Started" configs > > > Hello everyone! > > I have a small question. Why would not SER start with "serctl start" > command, if I am using configurations from getting started document? > (im not talking about the first config, but the latter ones). > > Andrey. > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

2 4

[Serusers] Fifo port when UA registered from port other than 5060
by Arek Bekiersz 28 Feb '06

28 Feb '06

I use SER fifo from a long time to initiate various requests. It works fine. But now I noticed that when UA is registered from port other than 5060 and I send some request to fifo (i.e. INVITE), this request will always be sent to port 5060.. SER is not doing any location lookup on that. Should I make registrar lookup myself, to check for port number that UA is registered. And only then send "dst_uri:PORT_NUMBER" to fifo? Because when I do, it works :-))) I use ser 0.8.12. Appreciate to have your advice on that. -- Regards, Arek Bekiersz

1 0

[Users] UDP Fragmentation (again)
by Helge Waastad 28 Feb '06

28 Feb '06

Hi, I've just had my first fragmentation problem. Since I'm using rr on my dispatchers and my proxies, INVITES from a gateway (i.e cisco) gets too large and are being fragmented...and of course not very good for nat'ed udp clients. Are there any other options than to remove_hf of less important headers like Remote-Party-ID and/or User-Agent? I guess TCP would be an option.... br hw -- Helge Waastad Senior Konsulent Systemavdelingen Smartnet

3 4

RE: [Users] Using t_reply()
by Douglas Garstang 28 Feb '06

28 Feb '06

You know, I have nightmares about OpenSER. Really. Ok... I kinda thought I had something working. I had this... It works great when a Decline comes back from 192.168.10.10. The failure_route block sees a negative reply, and sends Service Unavailable back to the phone... great... no problem. However, when there is NO REPLY from 192.168.10.10, t_check_status() STILL returns true. GOD OH GOD WHY? The docs say "in on_failure block - the code of the selected negative final reply".... well I didn't get a reply! So why does t_check_status() return true? Sorry for the emotion, but I've been dealing with OpenSER for 6 months now and I've never felt so much stress with any piece of software as I have with this one. Ever time I open the config file, I want to scream! route[10] { t_on_failure("11"); rewritehostport("192.168.10.10:5060"); if ( !t_relay() ) { xlog ("L_INFO","$avp(s:callid)[10]: t_relay() returned error"); } else { xlog ("L_INFO","$avp(s:callid)[10]: t_relay() returned ok"); } } failure_route[11] { route(12); if (t_check_status("")) { t_reply("503","Service Unavailable"); return; } else { route(12); } } -----Original Message----- From: Douglas Garstang Sent: Monday, February 27, 2006 6:24 PM To: users(a)openser.org Subject: [Users] Using t_reply() Why can't I do this??? Arrgh! I want to be able to call onreply_route[35] when we get a reply. So, not surprisingly, in onreply_route[35] I want to see what sort of reply I got, and then send ANOTHER reply back to the originater of the message telling them something was wrong. Why can't I use t_reply in a reply_route? What's the most likely away around this, because I sure must be missing something fundamental. route[10] { t_on_failure("11"); t_on_reply("35"); rewritehostport("pbx1-mgt0.ipt.oneeighty.com:5060"); if ( !t_relay() ) { xlog ("L_INFO","t_relay() returned error"); } else { xlog ("L_INFO","t_relay() returned ok"); } } onreply_route[35] { xlog ("L_INFO","Got some sort of reply"); if (t_check_status("603")) { xlog ("L_INFO","got a decline"); t_reply("603","Declined"); } return; } _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

2 1

[Serusers] Problem about 483 "too many hops"
by 牛坤 28 Feb '06

28 Feb '06

I recently installed ser 0.9.6 on my Sarge system.The server works fine when there's no www authentication.But after I setup my dns and add my domain in mysql's domain table, my ua always receives 483 error.After reading the following words, I add alias in my config file.But still it doesn't work. In both cases, the reason is probably an error in request routing script which caused an infinite loop. You can easily verify whether this happens by watching SIP traffic on loopback interface. A typical reason for misrouting is a failure to match local domain correctly. If a server fails to recognize a request for itself, it will try to forward it to current URI in believe it would forward them to a foreign domain. Alas, it forwards the request to itself again. This continues to happen until value of max_forwards header field reaches zero or the request grows too big. Solutions is easy: make sure that domain matching is correctly configured. A quick way to achieve that is to introduce a config option to ser.cfg: alias=domainname, where domainname shall be replaced with name of domain, which you wish to server and which appears Will someone be kind enough to make some useful suggestions?Any help would be appreciated.Thanks in advance.

1 0

[Serusers] how to set up ser as a proxy to monitor/forward all communication between users?
by lin9k＠yahoo.com 27 Feb '06

27 Feb '06

Hi, Anyone know how I can setup this function? If company policy need us to monitor all communication content (after successful SIP communication 200 OK) between users how can I do it in ser? Thanks in advance, ling __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] OpenSER v1.0.1 released
by Daniel-Constantin Mierla 27 Feb '06

27 Feb '06

Hello everybody, version 1.0.1 of OpenSER has been released. It is an update of version 1.0.0 and includes the fixes for the issues discovered in version 1.0.0, therefore all people using 1.0.0 should upgrade to this new version. The compatibility of the configuration file and database has been preserved (you do not need to do database reinstallation). Some details about v1.0.1 can be found at: http://www.openser.org/release-1.0.x.php Detailed changelog: http://www.openser.org/ChangeLog-1.0.1.php Sources as well as packages for different OSes can be downloaded from: http://openser.org/pub/openser/1.0.1/ The documentation for modules is posted at: http://openser.org/docs/modules/1.0.x/ If someone what to contribute with new packages, please send me the link from where I can download them. Cheers, Daniel

1 0

[Users] Errors on Starting TM and ACC
by Edson Gellert Schubert 27 Feb '06

27 Feb '06

Hi, list... We are migrating from SER 0.9.2 to OpenSER 1.0.0 The machine is a x86_64 Xeon 3.0GHz Dell, with ! GB RAM and LVM over a 70 SCSI HD. We are using OpenSUSE 10.0-OSS, with minimun packet selection, just added with OpenSER pre-reqs and nGrep. I run two versions of MySQL: the OpenSUSE stock 4.1.13, and 5.0.18, downloaded from MySQL site and compiled with 'rpmbuild'. GCC version please, see below on <0>. OpenSER compilations where all made with the command lines showed on <1>: I find many problems, not only during the compilation process, some are: 1- 'hiding' comments for RADIUS support on ACC (thanks Daniel for the help); 2- Makefile.defs:544: You are using an old and unsupported gcc version (4.0), compile at your own risk! (openser-1.0.0.-tls.tar.gz - stable) 3- '-mtune' value changed from 'x68_64' to 'x86-64' on Makefile.defs (CVS- snapshot); 4- an 'ld' message that I thing is a MySQL problem when using MySQL 5.0.18 libs (message, please see below on <2>) For the last issue I downgraded the MySQL to the stock version (4.1.13) but now I'm facing an runtime ERROR message. See below, on <3> the message from an "openser -TDdd" race. Any ideas? If the config file helps, please let me now. Edson. *<0>********************************************************************* #gcc -v Using built-in specs. Target: x86_64-suse-linux Configured with: ../configure --enable-threads=posix --prefix=/usr --with-local-prefix=/usr/local --infodir=/usr/share/info --mandir=/usr/share/man --libdir=/usr/lib64 --libexecdir=/usr/lib64 --enable-languages=c,c++,objc,f95,java,ada --disable-checking --with-gxx-include-dir=/usr/include/c++/4.0.2 --enable-java-awt=gtk --disable-libjava-multilib --with-slibdir=/lib64 --with-system-zlib --enable-shared --enable-__cxa_atexit --without-system-libunwind --host=x86_64-suse-linux Thread model: posix gcc version 4.0.2 20050901 (prerelease) (SUSE Linux) # *<1>********************************************************************* # make cfg-target=/etc/openser/ prefix=/usr cfg-prefix= # make cfg-target=/etc/openser/ prefix=/usr exclude_modules="postgres" \ > modules cfg-prefix= # make cfg-target=/etc/openser/ prefix=/usr exclude_modules="postgres" \ > install cfg-prefix= *<2>********************************************************************* /usr/lib64/gcc/x86_64-suse-linux/4.0.2/../../../../x86_64-suse-linux/bin/ld: /usr/lib64/mysql/libmysqlclient.a(libmysql.o): relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC /usr/lib64/mysql/libmysqlclient.a: could not read symbols: Bad value collect2: ld returned 1 exit status make[1]: *** [mysql.so] Error 1 make[1]: Leaving directory `/home/download/openser-1.0.0-tls/modules/mysql' *<3>********************************************************************* sip:/root # openser -TDdd Listening on udp: XXX.XXX.XXX.XXX [XXX.XXX.XXX.XXX]:5070 Aliases: *: sip.xxxx.com.br:* *: xxxx.com.br:* 0(31517) fm_malloc_init: F_OPTIMIZE=16384, /ROUNDTO=2048 0(31517) fm_malloc_init: F_HASH_SIZE=2099, fm_block size=33608 0(31517) fm_malloc_init(0x2aaaacebd000, 33554432), start=0x2aaaacebd000 0(31517) shm_mem_init: success 0(0) core dump limits set to 18446744073709551615 0(0) DEBUG: init_mod: mysql 0(0) mysql: MySQL client version is 4.1.13 0(0) DEBUG: init_mod: acc 0(0) ACC - initializing 0(0) find_export: found <load_tm> in module tm [/usr/lib/openser/modules/tm.so] 0(0) ERROR:tm:register_tmcb: callback type TMCB_REQUEST_IN registration attempt before TM module initialization 0(0) ERROR:acc:mod_init: cannot register TMCB_REQUEST_IN callback 0(0) init_mod(): Error while initializing module acc ERROR: error while initializing modules 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(0) destroy_tls: Entered 0(31519) shm_mem_destroy 0(31519) destroying the shared memory lock sip:/root #

2 1

[Users] how to log IP addresses from RTP
by Nathan Hawkins 27 Feb '06

27 Feb '06

I'm trying to figure out how to log RTP addresses from the SDP. I'd like to use db_extra to save the RTP endpoints from the INVITE and 200 messages. How can I get the addresses out of the SDP body and into AVP's? Also, if I'm using the TM module, can I still catch the 200 response and extract the response RTP IP address? Thanks, ---Nathan

2 1

[Serusers] Pb With Pstn-Gw
by Arnaud 27 Feb '06

27 Feb '06

Hi I've a pstn Gateway with two network cards, so i have 2 ips on these pstn gw One Ip, is used only for sip signalisation (sig) the second ip is only used for media stream The problem is: in the sdp headers: the sig ip is located in " o= " field (ex:o=192.168.0.1 ) and the media ip is located in " c= " field (ex:c=192.168.0.2) It works with some ip Phones or SoftPhone (ex: Xlite) But Not all the phones seems to read th "c=" Field, so the phones don't receive the media streams Is There Way or Workaround for theses phones to work with Ser ? My Ser Config is the Onsip Config Thank you very Much Regards arno

1 0

[Serusers] Error checking
by sip 27 Feb '06

27 Feb '06

I may have asked this last week, but if I did, I apologise. I have a rewriteuri call in SER. It essentially forwards all the packets to a different IP/port. Sometimes, when calling, everything goes through. Sometimes, that network is congested or confused, and I get a 404. I'm wondering if there's a good, clean way to handle those errors... is there some sort of check status or some such I could use to handle errors from a particular destination to which I forward certain calls? N.

2 1

[Users] Errors compiling against MySQL 5.0.18
by Edson 27 Feb '06

27 Feb '06

Hi, list... We are migrating from SER 0.9.2 to OpenSER 1.0.0 The machine is a x86_64 Xeon 3.0GHz Dell, with ! GB RAM and LVM over a 70 SCSI HD. We are using OpenSUSE 10.0-OSS, with minimun packet selection, just added with OpenSER pre-reqs and nGrep. I run two versions of MySQL: the OpenSUSE stock 4.1.13, and 5.0.18, downloaded from MySQL site and compiled with 'rpmbuild'. GCC version please, see below on <0>. OpenSER compilations where all made with the command lines showed on <1>: I find many problems, not only during the compilation process, some are: 1- 'hiding' comments for RADIUS support on ACC (thanks Daniel for the help); 2- Makefile.defs:544: You are using an old and unsupported gcc version (4.0), compile at your own risk! (openser-1.0.0.-tls.tar.gz - stable) 3- '-mtune' value had to be changed from 'x68_64' to 'x86-64' on Makefile.defs (CVS-snapshot) to compile; 4- an 'ld' message that I thing is a MySQL problem when using MySQL 5.0.18 libs (message, please see below on <2>) Radius is still given me some work, but I whant to use MySQL 5.x series. So, any one with a successful story in using CVS-head (or stable) with MySQL 5.0.x? Edson. *<0>********************************************************************* #gcc -v Using built-in specs. Target: x86_64-suse-linux Configured with: ../configure --enable-threads=posix --prefix=/usr --with-local-prefix=/usr/local --infodir=/usr/share/info --mandir=/usr/share/man --libdir=/usr/lib64 --libexecdir=/usr/lib64 --enable-languages=c,c++,objc,f95,java,ada --disable-checking --with-gxx-include-dir=/usr/include/c++/4.0.2 --enable-java-awt=gtk --disable-libjava-multilib --with-slibdir=/lib64 --with-system-zlib --enable-shared --enable-__cxa_atexit --without-system-libunwind --host=x86_64-suse-linux Thread model: posix gcc version 4.0.2 20050901 (prerelease) (SUSE Linux) # *<1>********************************************************************* # make cfg-target=/etc/openser/ prefix=/usr cfg-prefix= # make cfg-target=/etc/openser/ prefix=/usr exclude_modules="postgres" \ > modules cfg-prefix= # make cfg-target=/etc/openser/ prefix=/usr exclude_modules="postgres" \ > install cfg-prefix= *<2>********************************************************************* /usr/lib64/gcc/x86_64-suse-linux/4.0.2/../../../../x86_64-suse-linux/bin/ld: /usr/lib64/mysql/libmysqlclient.a(libmysql.o): relocation R_X86_64_32 against `a local symbol' can not be used when making a shared object; recompile with -fPIC /usr/lib64/mysql/libmysqlclient.a: could not read symbols: Bad value collect2: ld returned 1 exit status make[1]: *** [mysql.so] Error 1 make[1]: Leaving directory `/home/download/openser-1.0.0-tls/modules/mysql'

2 2

[Serusers] Serweb question
by frankie chiang 27 Feb '06

27 Feb '06

Dear all, Where is my mail setting for registration ? Is configure under config.php ? Where is the list of un-process registration ? How can I debug ? My serweb send a email but I can't receive it. Best Regard, __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] tls, xlog and select framework syntax
by Klaus Darilion 27 Feb '06

27 Feb '06

Hi! I want to log some TLS parameters. I've tried: xlog("L_ERR","@tls = %@tls (String description of the TLS layer)\n"); but all I get is: ser[20222]: @tls = <null>tls (String description of the TLS layer) using avps it works: %avp1=@tls; print_sattr("avp1"); xlog("L_ERR","avp1 = %$avp1\n"); What is the correct syntax for using the select framework? thanks klaus

2 6

[Serusers] "Getting Started" configs
by Andrey Kouprianov 27 Feb '06

27 Feb '06

Hello everyone! I have a small question. Why would not SER start with "serctl start" command, if I am using configurations from getting started document? (im not talking about the first config, but the latter ones). Andrey.

1 0

[Users] packaging v1.0.1
by Daniel-Constantin Mierla 27 Feb '06

27 Feb '06

Hello, the sources in branch rel_1_0_0 are ready to be packaged for release 1.0.1. In the "sip-server/packaging" most of the files were updated to the version 1.0.1 but some of them still need to be updated - I leave this for the one which generates the package because he knows better how to do it for specific OSes and architectures. Everyone which want to contribute with some packages is wellcome. I can provide some guidelines for generating rpms. For those which have no developer access to cvs, I posted and tarball with the latest sources at: http://openser.org/downloads/tmp/openser-1.0.1_src.tar.gz Once you generated the package, please let me know the link from where I can download. I will upload afterwards on openser.org site. Cheers, Daniel

1 1

[Serusers] SDP of 180
by Ever Zalazar 27 Feb '06

27 Feb '06

Hi people. I want to know if it's possible to configure SER to modify the SDP of the 180. Best Regards Ever Zalazar

1 0

[Serusers] how to rewrite radius request information packet before sending to radius
by gokhan uslu 27 Feb '06

27 Feb '06

Hi, I have a ser system talking with radius. there is no problem with authentication. but what i realized is that, in my radius request packet: USER-NAME= user1@ip_of_ser and this radius packet goes to radius for authentication. But i want to change the radius packet. so that my USER-NAME = user1@ip_of_ser will change to USER-NAME = user1@ip_of_user1 I need to change the authentication information of the radius packet, before sending it to radius. how can i handle this? do i need avp module or what? please help me? please --------------------------------- Yahoo! Mail Use Photomail to share photos without annoying attachments.

1 0

RE: [Users] nathelper natping OPTIONS packets formated to not get reply?
by Glenn Dalgliesh 27 Feb '06

27 Feb '06

Well I am feeling a little embarrassed at this point but it seems that my data may not have been exactly correct with regard to natpings not working without username in the TO field of OPTIONS packets. First my original test scenario was to ngrep for OPTIONS packets generated from OpenSer and see if I was getting response without Username in the TO field. Then I would send an OPTIONS using sipsak thru openser and see if I got a reply. Based on this test it appeared that it was the lack of Username in the TO field. But then after recording results with my work round in place which rewrote the received field. I found that I was still not getting consistent results on replies to OPTIONS packets this lead me to start looking to see if the OPTIONS packets where ever reaching the dest... It turns out that the OPTIONS packets were not reaching the dest and it seems that the reason was that the Cisco router btw OPENSER and the Internet was dropping a very high percentage of out bound packets under bursty situations such as hundreds of OPTIONS packets being sent to it at once. I have resolved this issues and now I am getting consistant results from the natping OPTIONS packets. I will follow up with one more test which is to remove my work around related to the received field and retest the results. However, I will have to wait until I can test this after hours inorder to make this change. Thanks and sorry for the bad data but as you can see it wasn't do to lack of thought :) FYI: your patch did seem to added the natping FROM field to the TO field after I retest we can figure out if that is really necessary.

2 3

[Users] forward_tcp
by Ramin Dousti 26 Feb '06

26 Feb '06

Hello, I'm trying to forward a SIP message that's been received by UDP, to $td through TCP. but the following configuration doesn't work: send_tcp("$td", 5060 ); or send_tcp($td, 5060 ); or avp_printf("s:dst", "$td"); send_tcp("s:dst", 5060 ); Could you please help? -- Ramin

3 3

[Serusers] lookup(): '' Not found in usrloc
by Kim Culhan 26 Feb '06

26 Feb '06

A UA is trying to send NOTIFY: U 1.2.3.4:56839 -> 5.6.7.8:5060 NOTIFY sip:machine.foo.com SIP/2.0. Via: SIP/2.0/UDP 1.2.3.4:56839;branch=z9hG4bK-ee80efbb. From: TPA - PSTN <sip:9876@machine.foo.com>;tag=ddf83b1b7d0b3b06o1. To: <sip:machine.foo.com>. Call-ID: e368f273-a2f62b8e(a)192.168.251.10. CSeq: 31511 NOTIFY. Max-Forwards: 70. Event: keep-alive. User-Agent: Sipura/SPA3000-2.0.13(GWg). Content-Length: 0. The reponse is: U 5.6.7.8:5060 -> 1.2.3.4:56839 SIP/2.0 404 User Not Found - lookup_location. Via: SIP/2.0/UDP 1.2.3.4:56839;branch=z9hG4bK-ee80efbb. From: TPA - PSTN <sip:9876@machine.foo.com>;tag=ddf83b1b7d0b3b06o1. To: <sip:machine.foo.com>;tag=efa044f399b3f75e8ec9d81a905f914f.d18f. Call-ID: e368f273-a2f62b8e(a)192.168.251.10. CSeq: 31511 NOTIFY. Server: Sip EXpress router (0.9.7-pre1 (i386/freebsd)). Content-Length: 0. Warning: 392 5.6.7.8:5060 "Noisy feedback tells: pid=15425 req_src_ip=1.2.3.4 req_src_port=56839 in_uri=sip:machine.foo.com out_uri=sip:machine.foo.com via_cnt==1". >From the logs we see among other things: Feb 25 08:09:13 machine /usr/local/sbin/ser[15425]: lookup(): '' Not found in usrloc Can anyone tell me why the user is not found ? Any help here is greatly appreciated -kim -- w8hdkim(a)gmail.com

6 8

[Serusers] problem with OpenSER + Asterisk + F5 load-balancing switch
by Jack Wei 26 Feb '06

26 Feb '06

Hi, I'm having a problem trying to set up OpenSER + Asterisk + F5 load-balancing switch. Establishing the call (w/ no audio) seems to work alright but when the client hangs up, the OK reply from Asterisk never reaches the client. By the way, I'll get to the audio problem after the SIP problem is solved. Here's my setup: 1) 2 OpenSER boxes behind NAT with 198.168.2.X IPs and a public virtual IP X.X.X.X. 2) 2 Asterisk boxes behind NAT with 192.168.3.X IPs and a public virtual IP Y.Y.Y.Y. 3) 2 MySQL database boxes behind NAT with 192.168.4.X IPs and a private virtual IP 192.168.4.10. Each set of boxes are in it's on VLAN so the gateways are .1 for each subnet. Attached are the logs from the private and public interfaces on the F5 switch. Can someone tell me what I'm doing wrong? or if my setup needs to be changed? Thanks in advance. Jack No. Time Source Destination Protocol Info 30 8.539666 X.X.X.X 192.168.2.11 SIP/SDP Request: INVITE sip:*0@X.X.X.X:5060, with session description Frame 30 (776 bytes on wire, 776 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 23018 (23018), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:*0@X.X.X.X:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3a000072cb00000029 From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Contact: <sip:jack.wei@64.77.225.69> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 1 INVITE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 262 Content-Type: application/sdp Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3349798330 3349798330 IN IP4 64.77.225.69 Session Name (s): SJphone Connection Information (c): IN IP4 64.77.225.69 Time Description, active time (t): 0 0 Session Attribute (a): setup:active Media Description, name and address (m): audio 49156 RTP/AVP 3 0 8 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16 No. Time Source Destination Protocol Info 31 8.540905 192.168.2.11 X.X.X.X SIP Status: 407 Proxy Authentication Required Frame 31 (598 bytes on wire, 598 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 23018 (23018) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header Via: SIP/2.0/UDP 10.1.201.69;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000072cb00000029;received=X.X.X.X From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=329cfeaa6ded039da25ff8cbb8668bd2.c651 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="X.X.X.X", nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd" Server: OpenSer (1.0.0-tls (i386/linux)) Content-Length: 0 No. Time Source Destination Protocol Info 35 8.543577 X.X.X.X 192.168.2.11 SIP Request: ACK sip:*0@X.X.X.X:5060 Frame 35 (479 bytes on wire, 479 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 23018 (23018), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:*0@X.X.X.X:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3a000072cb00000029 From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=329cfeaa6ded039da25ff8cbb8668bd2.c651 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 1 ACK Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 No. Time Source Destination Protocol Info 36 8.544953 X.X.X.X 192.168.2.11 SIP/SDP Request: INVITE sip:*0@X.X.X.X:5060, with session description Frame 36 (980 bytes on wire, 980 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 23018 (23018), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:*0@X.X.X.X:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Contact: <sip:jack.wei@64.77.225.69> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 262 Content-Type: application/sdp Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3349798330 3349798330 IN IP4 64.77.225.69 Session Name (s): SJphone Connection Information (c): IN IP4 64.77.225.69 Time Description, active time (t): 0 0 Session Attribute (a): setup:active Media Description, name and address (m): audio 49156 RTP/AVP 3 0 8 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16 No. Time Source Destination Protocol Info 139 8.859768 192.168.2.11 X.X.X.X SIP Status: 100 trying -- your call is important to us Frame 139 (461 bytes on wire, 461 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 23018 (23018) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Message Header Via: SIP/2.0/UDP 10.1.201.69;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a;received=X.X.X.X From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE Server: OpenSer (1.0.0-tls (i386/linux)) Content-Length: 0 No. Time Source Destination Protocol Info 140 8.859951 192.168.2.11 Y.Y.Y.Y SIP/SDP Request: INVITE sip:*02139435245@Y.Y.Y.Y:5060, with session description Frame 140 (1096 bytes on wire, 1096 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:*02139435245@Y.Y.Y.Y:5060 SIP/2.0 Message Header Record-Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKcee6.d2008e95.0 Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Contact: <sip:jack.wei@64.77.225.69> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE Max-Forwards: 69 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 262 Content-Type: application/sdp Remote-Party-ID: <sip:2139435245@X.X.X.X>;party=calling;id-type=subscriber;screen=yes;privacy=full P-hint: Asterisk outbound (voicemail box) Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3349798330 3349798330 IN IP4 64.77.225.69 Session Name (s): SJphone Connection Information (c): IN IP4 64.77.225.69 Time Description, active time (t): 0 0 Session Attribute (a): setup:active Media Description, name and address (m): audio 49156 RTP/AVP 3 0 8 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16 No. Time Source Destination Protocol Info 141 8.861041 192.168.2.1 192.168.2.11 SIP Status: 100 Trying Frame 141 (608 bytes on wire, 608 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: 192.168.2.1 (192.168.2.1), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Message Header Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKcee6.d2008e95.0;received=Y.Y.Y.Y Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@Y.Y.Y.Y> Content-Length: 0 No. Time Source Destination Protocol Info 143 8.861363 192.168.2.1 192.168.2.11 SIP/SDP Status: 200 OK, with session description Frame 143 (964 bytes on wire, 964 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: 192.168.2.1 (192.168.2.1), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKcee6.d2008e95.0;received=Y.Y.Y.Y Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a Record-Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@Y.Y.Y.Y> Content-Type: application/sdp Content-Length: 241 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 15210 15210 IN IP4 Y.Y.Y.Y Session Name (s): session Connection Information (c): IN IP4 Y.Y.Y.Y Time Description, active time (t): 0 0 Media Description, name and address (m): audio 13086 RTP/AVP 3 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): silenceSupp:off - - - - No. Time Source Destination Protocol Info 144 8.862363 192.168.2.11 X.X.X.X SIP/SDP Status: 200 OK, with session description Frame 144 (880 bytes on wire, 880 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 23018 (23018) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a Record-Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@Y.Y.Y.Y> Content-Type: application/sdp Content-Length: 241 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 15210 15210 IN IP4 Y.Y.Y.Y Session Name (s): session Connection Information (c): IN IP4 Y.Y.Y.Y Time Description, active time (t): 0 0 Media Description, name and address (m): audio 13086 RTP/AVP 3 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): silenceSupp:off - - - - No. Time Source Destination Protocol Info 148 8.867099 X.X.X.X 192.168.2.11 SIP Request: ACK sip:*02139435245@Y.Y.Y.Y Frame 148 (517 bytes on wire, 517 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 23018 (23018), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3b00004fe30000002d From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 ACK Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> No. Time Source Destination Protocol Info 166 8.871652 192.168.2.11 Y.Y.Y.Y SIP Request: ACK sip:*02139435245@Y.Y.Y.Y Frame 166 (624 bytes on wire, 624 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> Via: SIP/2.0/UDP 192.168.2.11;branch=0 Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3b00004fe30000002d From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 ACK Max-Forwards: 69 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 P-hint: rr-enforced No. Time Source Destination Protocol Info 182 10.744014 X.X.X.X 192.168.2.11 SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 182 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 23018 (23018), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 183 10.744631 192.168.2.11 Y.Y.Y.Y SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 183 (849 bytes on wire, 849 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKdee6.210f3e03.0 Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 69 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" P-hint: rr-enforced No. Time Source Destination Protocol Info 184 10.745306 192.168.2.1 192.168.2.11 SIP Status: 200 OK Frame 184 (697 bytes on wire, 697 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: 192.168.2.1 (192.168.2.1), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKdee6.210f3e03.0;received=Y.Y.Y.Y Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@Y.Y.Y.Y> Content-Length: 0 No. Time Source Destination Protocol Info 186 10.746409 192.168.2.11 X.X.X.X SIP Status: 200 OK Frame 186 (639 bytes on wire, 639 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 23018 (23018) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@192.168.2.1:5060> Content-Length: 0 P-hint: NAT traversal No. Time Source Destination Protocol Info 277 11.244153 X.X.X.X 192.168.2.11 SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 277 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 291 11.248051 192.168.2.11 X.X.X.X SIP Status: 200 OK Frame 291 (639 bytes on wire, 639 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 23018 (23018) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@192.168.2.1:5060> Content-Length: 0 P-hint: NAT traversal No. Time Source Destination Protocol Info 357 12.243451 X.X.X.X 192.168.2.11 SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 357 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 358 12.244007 192.168.2.11 X.X.X.X SIP Status: 200 OK Frame 358 (639 bytes on wire, 639 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 23018 (23018) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@192.168.2.1:5060> Content-Length: 0 P-hint: NAT traversal No. Time Source Destination Protocol Info 430 14.241814 X.X.X.X 192.168.2.11 SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 430 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 431 14.242562 192.168.2.11 X.X.X.X SIP Status: 200 OK Frame 431 (639 bytes on wire, 639 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 23018 (23018) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@192.168.2.1:5060> Content-Length: 0 P-hint: NAT traversal No. Time Source Destination Protocol Info 508 18.247779 X.X.X.X 192.168.2.11 SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 508 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 509 18.248381 192.168.2.11 Y.Y.Y.Y SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 509 (848 bytes on wire, 848 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKdee6.310f3e03.0 Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=5060;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 69 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" P-hint: rr-enforced No. Time Source Destination Protocol Info 510 18.249048 192.168.2.1 192.168.2.11 SIP Status: 200 OK Frame 510 (696 bytes on wire, 696 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: 192.168.2.1 (192.168.2.1), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKdee6.310f3e03.0;received=Y.Y.Y.Y Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=5060;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@Y.Y.Y.Y> Content-Length: 0 No. Time Source Destination Protocol Info 511 18.249538 192.168.2.11 X.X.X.X SIP Status: 200 OK Frame 511 (638 bytes on wire, 638 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=5060;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@192.168.2.1:5060> Content-Length: 0 P-hint: NAT traversal No. Time Source Destination Protocol Info 512 18.249612 X.X.X.X 192.168.2.11 SIP Status: 200 OK Frame 512 (638 bytes on wire, 638 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 23036 (23036), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=5060;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@192.168.2.1:5060> Content-Length: 0 P-hint: NAT traversal No. Time Source Destination Protocol Info 13 9.169758 64.77.225.69 X.X.X.X SIP/SDP Request: INVITE sip:*0@X.X.X.X:5060, with session description Frame 13 (776 bytes on wire, 776 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:*0@X.X.X.X:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3a000072cb00000029 From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Contact: <sip:jack.wei@64.77.225.69> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 1 INVITE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 262 Content-Type: application/sdp Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3349798330 3349798330 IN IP4 64.77.225.69 Session Name (s): SJphone Connection Information (c): IN IP4 64.77.225.69 Time Description, active time (t): 0 0 Session Attribute (a): setup:active Media Description, name and address (m): audio 49156 RTP/AVP 3 0 8 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16 No. Time Source Destination Protocol Info 14 9.171073 X.X.X.X 64.77.225.69 SIP Status: 407 Proxy Authentication Required Frame 14 (598 bytes on wire, 598 bytes captured) Ethernet II, Src: 64.77.239.162 (00:01:d7:37:0e:c3), Dst: Cisco_20:a3:2c (00:02:7e:20:a3:2c) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 64.77.225.69 (64.77.225.69) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header Via: SIP/2.0/UDP 10.1.201.69;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000072cb00000029;received=X.X.X.X From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=329cfeaa6ded039da25ff8cbb8668bd2.c651 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="X.X.X.X", nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd" Server: OpenSer (1.0.0-tls (i386/linux)) Content-Length: 0 No. Time Source Destination Protocol Info 16 9.173686 64.77.225.69 X.X.X.X SIP Request: ACK sip:*0@X.X.X.X:5060 Frame 16 (479 bytes on wire, 479 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:*0@X.X.X.X:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3a000072cb00000029 From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=329cfeaa6ded039da25ff8cbb8668bd2.c651 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 1 ACK Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 No. Time Source Destination Protocol Info 17 9.175053 64.77.225.69 X.X.X.X SIP/SDP Request: INVITE sip:*0@X.X.X.X:5060, with session description Frame 17 (980 bytes on wire, 980 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:*0@X.X.X.X:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Contact: <sip:jack.wei@64.77.225.69> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 262 Content-Type: application/sdp Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3349798330 3349798330 IN IP4 64.77.225.69 Session Name (s): SJphone Connection Information (c): IN IP4 64.77.225.69 Time Description, active time (t): 0 0 Session Attribute (a): setup:active Media Description, name and address (m): audio 49156 RTP/AVP 3 0 8 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16 No. Time Source Destination Protocol Info 20 9.489937 X.X.X.X 64.77.225.69 SIP Status: 100 trying -- your call is important to us Frame 20 (461 bytes on wire, 461 bytes captured) Ethernet II, Src: 64.77.239.162 (00:01:d7:37:0e:c3), Dst: Cisco_20:a3:2c (00:02:7e:20:a3:2c) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 64.77.225.69 (64.77.225.69) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Message Header Via: SIP/2.0/UDP 10.1.201.69;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a;received=X.X.X.X From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE Server: OpenSer (1.0.0-tls (i386/linux)) Content-Length: 0 No. Time Source Destination Protocol Info 21 9.492532 X.X.X.X 64.77.225.69 SIP/SDP Status: 200 OK, with session description Frame 21 (880 bytes on wire, 880 bytes captured) Ethernet II, Src: 64.77.239.162 (00:01:d7:37:0e:c3), Dst: Cisco_20:a3:2c (00:02:7e:20:a3:2c) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 64.77.225.69 (64.77.225.69) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a Record-Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@Y.Y.Y.Y> Content-Type: application/sdp Content-Length: 241 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 15210 15210 IN IP4 Y.Y.Y.Y Session Name (s): session Connection Information (c): IN IP4 Y.Y.Y.Y Time Description, active time (t): 0 0 Media Description, name and address (m): audio 13086 RTP/AVP 3 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): silenceSupp:off - - - - No. Time Source Destination Protocol Info 22 9.497201 64.77.225.69 X.X.X.X SIP Request: ACK sip:*02139435245@Y.Y.Y.Y Frame 22 (517 bytes on wire, 517 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3b00004fe30000002d From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 ACK Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> No. Time Source Destination Protocol Info 23 9.500885 64.77.225.69 Y.Y.Y.Y RTP Payload type=GSM 06.10, SSRC=247342095, Seq=1084, Time=0, Mark Frame 23 (87 bytes on wire, 87 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 49156 (49156), Dst Port: 13086 (13086) Real-Time Transport Protocol No. Time Source Destination Protocol Info 30 9.580797 64.77.225.69 Y.Y.Y.Y RTP Payload type=GSM 06.10, SSRC=247342095, Seq=1088, Time=640 Frame 30 (87 bytes on wire, 87 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 49156 (49156), Dst Port: 13086 (13086) Real-Time Transport Protocol No. Time Source Destination Protocol Info 31 9.600740 64.77.225.69 Y.Y.Y.Y RTP Payload type=GSM 06.10, SSRC=247342095, Seq=1089, Time=800 Frame 31 (87 bytes on wire, 87 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 49156 (49156), Dst Port: 13086 (13086) Real-Time Transport Protocol No. Time Source Destination Protocol Info 35 9.680737 64.77.225.69 Y.Y.Y.Y RTP Payload type=GSM 06.10, SSRC=247342095, Seq=1093, Time=1440 Frame 35 (87 bytes on wire, 87 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 49156 (49156), Dst Port: 13086 (13086) Real-Time Transport Protocol No. Time Source Destination Protocol Info 36 9.700744 64.77.225.69 Y.Y.Y.Y RTP Payload type=GSM 06.10, SSRC=247342095, Seq=1094, Time=1600 Frame 36 (87 bytes on wire, 87 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 49156 (49156), Dst Port: 13086 (13086) Real-Time Transport Protocol No. Time Source Destination Protocol Info 123 11.374114 64.77.225.69 X.X.X.X SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 123 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 129 11.874205 64.77.225.69 X.X.X.X SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 129 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 135 12.873507 64.77.225.69 X.X.X.X SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 135 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 139 13.492072 64.77.239.162 64.77.226.30 TCP 32800 > 3306 [SYN] Seq=0 Ack=0 Win=23360 Len=0 MSS=1460 TSV=682543378 TSER=0 WS=2 Frame 139 (74 bytes on wire, 74 bytes captured) Ethernet II, Src: 64.77.239.162 (00:01:d7:37:0e:c3), Dst: Cisco_20:a3:2c (00:02:7e:20:a3:2c) Internet Protocol, Src: 64.77.239.162 (64.77.239.162), Dst: 64.77.226.30 (64.77.226.30) Transmission Control Protocol, Src Port: 32800 (32800), Dst Port: 3306 (3306), Seq: 0, Ack: 0, Len: 0 No. Time Source Destination Protocol Info 140 13.624964 64.77.239.162 64.77.226.27 DNS Standard query PTR 1.4.168.192.in-addr.arpa Frame 140 (84 bytes on wire, 84 bytes captured) Ethernet II, Src: 64.77.239.162 (00:01:d7:37:0e:c3), Dst: Cisco_20:a3:2c (00:02:7e:20:a3:2c) Internet Protocol, Src: 64.77.239.162 (64.77.239.162), Dst: 64.77.226.27 (64.77.226.27) User Datagram Protocol, Src Port: 32771 (32771), Dst Port: domain (53) Domain Name System (query) No. Time Source Destination Protocol Info 141 13.625929 64.77.226.27 64.77.239.162 DNS Standard query response, No such name Frame 141 (161 bytes on wire, 161 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.226.27 (64.77.226.27), Dst: 64.77.239.162 (64.77.239.162) User Datagram Protocol, Src Port: domain (53), Dst Port: 32771 (32771) Domain Name System (response) No. Time Source Destination Protocol Info 143 14.031913 64.77.239.161 224.0.0.5 OSPF Hello Packet Frame 143 (78 bytes on wire, 78 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 01:00:5e:00:00:05 (01:00:5e:00:00:05) Internet Protocol, Src: 64.77.239.161 (64.77.239.161), Dst: 224.0.0.5 (224.0.0.5) Open Shortest Path First No. Time Source Destination Protocol Info 144 14.871869 64.77.225.69 X.X.X.X SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 144 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 148 14.975012 X.X.X.X 64.77.225.69 TCP 22 > 1321 [ACK] Seq=1588 Ack=52 Win=16384 Len=0 Frame 148 (54 bytes on wire, 54 bytes captured) Ethernet II, Src: 64.77.239.162 (00:01:d7:37:0e:c3), Dst: Cisco_20:a3:2c (00:02:7e:20:a3:2c) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 64.77.225.69 (64.77.225.69) Transmission Control Protocol, Src Port: 22 (22), Dst Port: 1321 (1321), Seq: 1588, Ack: 52, Len: 0 No. Time Source Destination Protocol Info 154 18.877837 64.77.225.69 X.X.X.X SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 154 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5"

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[Users] problem with OpenSER + Asterisk + F5 load-balancing switch
by Jack Wei 26 Feb '06

26 Feb '06

Hi, I'm having a problem trying to set up OpenSER + Asterisk + F5 load-balancing switch. Establishing the call (w/ no audio) seems to work alright but when the client hangs up, the OK reply from Asterisk never reaches the client. By the way, I'll get to the audio problem after the SIP problem is solved. Here's my setup: 1) 2 OpenSER boxes behind NAT with 198.168.2.X IPs and a public virtual IP X.X.X.X. 2) 2 Asterisk boxes behind NAT with 192.168.3.X IPs and a public virtual IP Y.Y.Y.Y. 3) 2 MySQL database boxes behind NAT with 192.168.4.X IPs and a private virtual IP 192.168.4.10. Each set of boxes are in it's on VLAN so the gateways are .1 for each subnet. Attached are the logs from the private and public interfaces on the F5 switch. Can someone tell me what I'm doing wrong? or if my setup needs to be changed? Thanks in advance. Jack No. Time Source Destination Protocol Info 13 9.169758 64.77.225.69 X.X.X.X SIP/SDP Request: INVITE sip:*0@X.X.X.X:5060, with session description Frame 13 (776 bytes on wire, 776 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:*0@X.X.X.X:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3a000072cb00000029 From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Contact: <sip:jack.wei@64.77.225.69> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 1 INVITE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 262 Content-Type: application/sdp Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3349798330 3349798330 IN IP4 64.77.225.69 Session Name (s): SJphone Connection Information (c): IN IP4 64.77.225.69 Time Description, active time (t): 0 0 Session Attribute (a): setup:active Media Description, name and address (m): audio 49156 RTP/AVP 3 0 8 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16 No. Time Source Destination Protocol Info 14 9.171073 X.X.X.X 64.77.225.69 SIP Status: 407 Proxy Authentication Required Frame 14 (598 bytes on wire, 598 bytes captured) Ethernet II, Src: 64.77.239.162 (00:01:d7:37:0e:c3), Dst: Cisco_20:a3:2c (00:02:7e:20:a3:2c) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 64.77.225.69 (64.77.225.69) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header Via: SIP/2.0/UDP 10.1.201.69;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000072cb00000029;received=X.X.X.X From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=329cfeaa6ded039da25ff8cbb8668bd2.c651 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="X.X.X.X", nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd" Server: OpenSer (1.0.0-tls (i386/linux)) Content-Length: 0 No. Time Source Destination Protocol Info 16 9.173686 64.77.225.69 X.X.X.X SIP Request: ACK sip:*0@X.X.X.X:5060 Frame 16 (479 bytes on wire, 479 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:*0@X.X.X.X:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3a000072cb00000029 From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=329cfeaa6ded039da25ff8cbb8668bd2.c651 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 1 ACK Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 No. Time Source Destination Protocol Info 17 9.175053 64.77.225.69 X.X.X.X SIP/SDP Request: INVITE sip:*0@X.X.X.X:5060, with session description Frame 17 (980 bytes on wire, 980 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:*0@X.X.X.X:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Contact: <sip:jack.wei@64.77.225.69> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 262 Content-Type: application/sdp Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3349798330 3349798330 IN IP4 64.77.225.69 Session Name (s): SJphone Connection Information (c): IN IP4 64.77.225.69 Time Description, active time (t): 0 0 Session Attribute (a): setup:active Media Description, name and address (m): audio 49156 RTP/AVP 3 0 8 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16 No. Time Source Destination Protocol Info 20 9.489937 X.X.X.X 64.77.225.69 SIP Status: 100 trying -- your call is important to us Frame 20 (461 bytes on wire, 461 bytes captured) Ethernet II, Src: 64.77.239.162 (00:01:d7:37:0e:c3), Dst: Cisco_20:a3:2c (00:02:7e:20:a3:2c) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 64.77.225.69 (64.77.225.69) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Message Header Via: SIP/2.0/UDP 10.1.201.69;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a;received=X.X.X.X From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE Server: OpenSer (1.0.0-tls (i386/linux)) Content-Length: 0 No. Time Source Destination Protocol Info 21 9.492532 X.X.X.X 64.77.225.69 SIP/SDP Status: 200 OK, with session description Frame 21 (880 bytes on wire, 880 bytes captured) Ethernet II, Src: 64.77.239.162 (00:01:d7:37:0e:c3), Dst: Cisco_20:a3:2c (00:02:7e:20:a3:2c) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 64.77.225.69 (64.77.225.69) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a Record-Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@Y.Y.Y.Y> Content-Type: application/sdp Content-Length: 241 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 15210 15210 IN IP4 Y.Y.Y.Y Session Name (s): session Connection Information (c): IN IP4 Y.Y.Y.Y Time Description, active time (t): 0 0 Media Description, name and address (m): audio 13086 RTP/AVP 3 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): silenceSupp:off - - - - No. Time Source Destination Protocol Info 22 9.497201 64.77.225.69 X.X.X.X SIP Request: ACK sip:*02139435245@Y.Y.Y.Y Frame 22 (517 bytes on wire, 517 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3b00004fe30000002d From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 ACK Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> No. Time Source Destination Protocol Info 23 9.500885 64.77.225.69 Y.Y.Y.Y RTP Payload type=GSM 06.10, SSRC=247342095, Seq=1084, Time=0, Mark Frame 23 (87 bytes on wire, 87 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 49156 (49156), Dst Port: 13086 (13086) Real-Time Transport Protocol No. Time Source Destination Protocol Info 30 9.580797 64.77.225.69 Y.Y.Y.Y RTP Payload type=GSM 06.10, SSRC=247342095, Seq=1088, Time=640 Frame 30 (87 bytes on wire, 87 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 49156 (49156), Dst Port: 13086 (13086) Real-Time Transport Protocol No. Time Source Destination Protocol Info 31 9.600740 64.77.225.69 Y.Y.Y.Y RTP Payload type=GSM 06.10, SSRC=247342095, Seq=1089, Time=800 Frame 31 (87 bytes on wire, 87 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 49156 (49156), Dst Port: 13086 (13086) Real-Time Transport Protocol No. Time Source Destination Protocol Info 35 9.680737 64.77.225.69 Y.Y.Y.Y RTP Payload type=GSM 06.10, SSRC=247342095, Seq=1093, Time=1440 Frame 35 (87 bytes on wire, 87 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 49156 (49156), Dst Port: 13086 (13086) Real-Time Transport Protocol No. Time Source Destination Protocol Info 36 9.700744 64.77.225.69 Y.Y.Y.Y RTP Payload type=GSM 06.10, SSRC=247342095, Seq=1094, Time=1600 Frame 36 (87 bytes on wire, 87 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 49156 (49156), Dst Port: 13086 (13086) Real-Time Transport Protocol No. Time Source Destination Protocol Info 123 11.374114 64.77.225.69 X.X.X.X SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 123 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 129 11.874205 64.77.225.69 X.X.X.X SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 129 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 135 12.873507 64.77.225.69 X.X.X.X SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 135 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 139 13.492072 64.77.239.162 64.77.226.30 TCP 32800 > 3306 [SYN] Seq=0 Ack=0 Win=23360 Len=0 MSS=1460 TSV=682543378 TSER=0 WS=2 Frame 139 (74 bytes on wire, 74 bytes captured) Ethernet II, Src: 64.77.239.162 (00:01:d7:37:0e:c3), Dst: Cisco_20:a3:2c (00:02:7e:20:a3:2c) Internet Protocol, Src: 64.77.239.162 (64.77.239.162), Dst: 64.77.226.30 (64.77.226.30) Transmission Control Protocol, Src Port: 32800 (32800), Dst Port: 3306 (3306), Seq: 0, Ack: 0, Len: 0 No. Time Source Destination Protocol Info 140 13.624964 64.77.239.162 64.77.226.27 DNS Standard query PTR 1.4.168.192.in-addr.arpa Frame 140 (84 bytes on wire, 84 bytes captured) Ethernet II, Src: 64.77.239.162 (00:01:d7:37:0e:c3), Dst: Cisco_20:a3:2c (00:02:7e:20:a3:2c) Internet Protocol, Src: 64.77.239.162 (64.77.239.162), Dst: 64.77.226.27 (64.77.226.27) User Datagram Protocol, Src Port: 32771 (32771), Dst Port: domain (53) Domain Name System (query) No. Time Source Destination Protocol Info 141 13.625929 64.77.226.27 64.77.239.162 DNS Standard query response, No such name Frame 141 (161 bytes on wire, 161 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.226.27 (64.77.226.27), Dst: 64.77.239.162 (64.77.239.162) User Datagram Protocol, Src Port: domain (53), Dst Port: 32771 (32771) Domain Name System (response) No. Time Source Destination Protocol Info 143 14.031913 64.77.239.161 224.0.0.5 OSPF Hello Packet Frame 143 (78 bytes on wire, 78 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 01:00:5e:00:00:05 (01:00:5e:00:00:05) Internet Protocol, Src: 64.77.239.161 (64.77.239.161), Dst: 224.0.0.5 (224.0.0.5) Open Shortest Path First No. Time Source Destination Protocol Info 144 14.871869 64.77.225.69 X.X.X.X SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 144 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 148 14.975012 X.X.X.X 64.77.225.69 TCP 22 > 1321 [ACK] Seq=1588 Ack=52 Win=16384 Len=0 Frame 148 (54 bytes on wire, 54 bytes captured) Ethernet II, Src: 64.77.239.162 (00:01:d7:37:0e:c3), Dst: Cisco_20:a3:2c (00:02:7e:20:a3:2c) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 64.77.225.69 (64.77.225.69) Transmission Control Protocol, Src Port: 22 (22), Dst Port: 1321 (1321), Seq: 1588, Ack: 52, Len: 0 No. Time Source Destination Protocol Info 154 18.877837 64.77.225.69 X.X.X.X SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 154 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: Cisco_20:a3:2c (00:02:7e:20:a3:2c), Dst: 64.77.239.162 (00:01:d7:37:0e:c3) Internet Protocol, Src: 64.77.225.69 (64.77.225.69), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 30 8.539666 X.X.X.X 192.168.2.11 SIP/SDP Request: INVITE sip:*0@X.X.X.X:5060, with session description Frame 30 (776 bytes on wire, 776 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 23018 (23018), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:*0@X.X.X.X:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3a000072cb00000029 From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Contact: <sip:jack.wei@64.77.225.69> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 1 INVITE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 262 Content-Type: application/sdp Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3349798330 3349798330 IN IP4 64.77.225.69 Session Name (s): SJphone Connection Information (c): IN IP4 64.77.225.69 Time Description, active time (t): 0 0 Session Attribute (a): setup:active Media Description, name and address (m): audio 49156 RTP/AVP 3 0 8 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16 No. Time Source Destination Protocol Info 31 8.540905 192.168.2.11 X.X.X.X SIP Status: 407 Proxy Authentication Required Frame 31 (598 bytes on wire, 598 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 23018 (23018) Session Initiation Protocol Status-Line: SIP/2.0 407 Proxy Authentication Required Message Header Via: SIP/2.0/UDP 10.1.201.69;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000072cb00000029;received=X.X.X.X From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=329cfeaa6ded039da25ff8cbb8668bd2.c651 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="X.X.X.X", nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd" Server: OpenSer (1.0.0-tls (i386/linux)) Content-Length: 0 No. Time Source Destination Protocol Info 35 8.543577 X.X.X.X 192.168.2.11 SIP Request: ACK sip:*0@X.X.X.X:5060 Frame 35 (479 bytes on wire, 479 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 23018 (23018), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:*0@X.X.X.X:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3a000072cb00000029 From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=329cfeaa6ded039da25ff8cbb8668bd2.c651 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 1 ACK Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 No. Time Source Destination Protocol Info 36 8.544953 X.X.X.X 192.168.2.11 SIP/SDP Request: INVITE sip:*0@X.X.X.X:5060, with session description Frame 36 (980 bytes on wire, 980 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 23018 (23018), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:*0@X.X.X.X:5060 SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Contact: <sip:jack.wei@64.77.225.69> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 262 Content-Type: application/sdp Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3349798330 3349798330 IN IP4 64.77.225.69 Session Name (s): SJphone Connection Information (c): IN IP4 64.77.225.69 Time Description, active time (t): 0 0 Session Attribute (a): setup:active Media Description, name and address (m): audio 49156 RTP/AVP 3 0 8 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16 No. Time Source Destination Protocol Info 139 8.859768 192.168.2.11 X.X.X.X SIP Status: 100 trying -- your call is important to us Frame 139 (461 bytes on wire, 461 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 23018 (23018) Session Initiation Protocol Status-Line: SIP/2.0 100 trying -- your call is important to us Message Header Via: SIP/2.0/UDP 10.1.201.69;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a;received=X.X.X.X From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE Server: OpenSer (1.0.0-tls (i386/linux)) Content-Length: 0 No. Time Source Destination Protocol Info 140 8.859951 192.168.2.11 Y.Y.Y.Y SIP/SDP Request: INVITE sip:*02139435245@Y.Y.Y.Y:5060, with session description Frame 140 (1096 bytes on wire, 1096 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: INVITE sip:*02139435245@Y.Y.Y.Y:5060 SIP/2.0 Message Header Record-Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKcee6.d2008e95.0 Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Contact: <sip:jack.wei@64.77.225.69> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE Max-Forwards: 69 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 262 Content-Type: application/sdp Remote-Party-ID: <sip:2139435245@X.X.X.X>;party=calling;id-type=subscriber;screen=yes;privacy=full P-hint: Asterisk outbound (voicemail box) Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): - 3349798330 3349798330 IN IP4 64.77.225.69 Session Name (s): SJphone Connection Information (c): IN IP4 64.77.225.69 Time Description, active time (t): 0 0 Session Attribute (a): setup:active Media Description, name and address (m): audio 49156 RTP/AVP 3 0 8 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:8 PCMA/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-11,16 No. Time Source Destination Protocol Info 141 8.861041 192.168.2.1 192.168.2.11 SIP Status: 100 Trying Frame 141 (608 bytes on wire, 608 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: 192.168.2.1 (192.168.2.1), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 100 Trying Message Header Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKcee6.d2008e95.0;received=Y.Y.Y.Y Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060> Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@Y.Y.Y.Y> Content-Length: 0 No. Time Source Destination Protocol Info 143 8.861363 192.168.2.1 192.168.2.11 SIP/SDP Status: 200 OK, with session description Frame 143 (964 bytes on wire, 964 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: 192.168.2.1 (192.168.2.1), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKcee6.d2008e95.0;received=Y.Y.Y.Y Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a Record-Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@Y.Y.Y.Y> Content-Type: application/sdp Content-Length: 241 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 15210 15210 IN IP4 Y.Y.Y.Y Session Name (s): session Connection Information (c): IN IP4 Y.Y.Y.Y Time Description, active time (t): 0 0 Media Description, name and address (m): audio 13086 RTP/AVP 3 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): silenceSupp:off - - - - No. Time Source Destination Protocol Info 144 8.862363 192.168.2.11 X.X.X.X SIP/SDP Status: 200 OK, with session description Frame 144 (880 bytes on wire, 880 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 23018 (23018) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3a000003630000002a Record-Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@Y.Y.Y.Y> Content-Type: application/sdp Content-Length: 241 Message body Session Description Protocol Session Description Protocol Version (v): 0 Owner/Creator, Session Id (o): root 15210 15210 IN IP4 Y.Y.Y.Y Session Name (s): session Connection Information (c): IN IP4 Y.Y.Y.Y Time Description, active time (t): 0 0 Media Description, name and address (m): audio 13086 RTP/AVP 3 0 101 Media Attribute (a): rtpmap:3 GSM/8000 Media Attribute (a): rtpmap:0 PCMU/8000 Media Attribute (a): rtpmap:101 telephone-event/8000 Media Attribute (a): fmtp:101 0-16 Media Attribute (a): silenceSupp:off - - - - No. Time Source Destination Protocol Info 148 8.867099 X.X.X.X 192.168.2.11 SIP Request: ACK sip:*02139435245@Y.Y.Y.Y Frame 148 (517 bytes on wire, 517 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 23018 (23018), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3b00004fe30000002d From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 ACK Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> No. Time Source Destination Protocol Info 166 8.871652 192.168.2.11 Y.Y.Y.Y SIP Request: ACK sip:*02139435245@Y.Y.Y.Y Frame 166 (624 bytes on wire, 624 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: ACK sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> Via: SIP/2.0/UDP 192.168.2.11;branch=0 Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3b00004fe30000002d From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 2 ACK Max-Forwards: 69 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 P-hint: rr-enforced No. Time Source Destination Protocol Info 182 10.744014 X.X.X.X 192.168.2.11 SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 182 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 23018 (23018), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 183 10.744631 192.168.2.11 Y.Y.Y.Y SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 183 (849 bytes on wire, 849 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKdee6.210f3e03.0 Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 69 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" P-hint: rr-enforced No. Time Source Destination Protocol Info 184 10.745306 192.168.2.1 192.168.2.11 SIP Status: 200 OK Frame 184 (697 bytes on wire, 697 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: 192.168.2.1 (192.168.2.1), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKdee6.210f3e03.0;received=Y.Y.Y.Y Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@Y.Y.Y.Y> Content-Length: 0 No. Time Source Destination Protocol Info 186 10.746409 192.168.2.11 X.X.X.X SIP Status: 200 OK Frame 186 (639 bytes on wire, 639 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 23018 (23018) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@192.168.2.1:5060> Content-Length: 0 P-hint: NAT traversal No. Time Source Destination Protocol Info 277 11.244153 X.X.X.X 192.168.2.11 SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 277 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 291 11.248051 192.168.2.11 X.X.X.X SIP Status: 200 OK Frame 291 (639 bytes on wire, 639 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 23018 (23018) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@192.168.2.1:5060> Content-Length: 0 P-hint: NAT traversal No. Time Source Destination Protocol Info 357 12.243451 X.X.X.X 192.168.2.11 SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 357 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 358 12.244007 192.168.2.11 X.X.X.X SIP Status: 200 OK Frame 358 (639 bytes on wire, 639 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 23018 (23018) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@192.168.2.1:5060> Content-Length: 0 P-hint: NAT traversal No. Time Source Destination Protocol Info 430 14.241814 X.X.X.X 192.168.2.11 SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 430 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 431 14.242562 192.168.2.11 X.X.X.X SIP Status: 200 OK Frame 431 (639 bytes on wire, 639 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 23018 (23018) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=23018;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@192.168.2.1:5060> Content-Length: 0 P-hint: NAT traversal No. Time Source Destination Protocol Info 508 18.247779 X.X.X.X 192.168.2.11 SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 508 (721 bytes on wire, 721 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Via: SIP/2.0/UDP 10.1.201.69;rport;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 70 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Route: <sip:*0@X.X.X.X:5060;ftag=580759013014;lr=on> Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" No. Time Source Destination Protocol Info 509 18.248381 192.168.2.11 Y.Y.Y.Y SIP Request: BYE sip:*02139435245@Y.Y.Y.Y Frame 509 (848 bytes on wire, 848 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: Y.Y.Y.Y (Y.Y.Y.Y) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Request-Line: BYE sip:*02139435245@Y.Y.Y.Y SIP/2.0 Message Header Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKdee6.310f3e03.0 Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=5060;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE Max-Forwards: 69 User-Agent: SJphone/1.61.312b (SJ Labs) Content-Length: 0 Proxy-Authorization: Digest username="jack.wei",realm="X.X.X.X",nonce="43fefdd01e0923dcfc87e7d23d9b731cfc84ffbd",uri="sip:*0@X.X.X.X:5060",response="0b87bcb43183f39a2d19f11572b6c0c5" P-hint: rr-enforced No. Time Source Destination Protocol Info 510 18.249048 192.168.2.1 192.168.2.11 SIP Status: 200 OK Frame 510 (696 bytes on wire, 696 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: 192.168.2.1 (192.168.2.1), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 192.168.2.11;branch=z9hG4bKdee6.310f3e03.0;received=Y.Y.Y.Y Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=5060;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@Y.Y.Y.Y> Content-Length: 0 No. Time Source Destination Protocol Info 511 18.249538 192.168.2.11 X.X.X.X SIP Status: 200 OK Frame 511 (638 bytes on wire, 638 bytes captured) Ethernet II, Src: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b), Dst: 192.168.2.1 (00:01:d7:37:0e:c2) Internet Protocol, Src: 192.168.2.11 (192.168.2.11), Dst: X.X.X.X (X.X.X.X) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=5060;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@192.168.2.1:5060> Content-Length: 0 P-hint: NAT traversal No. Time Source Destination Protocol Info 512 18.249612 X.X.X.X 192.168.2.11 SIP Status: 200 OK Frame 512 (638 bytes on wire, 638 bytes captured) Ethernet II, Src: 192.168.2.1 (00:01:d7:37:0e:c2), Dst: 00:16:35:9f:2c:6b (00:16:35:9f:2c:6b) Internet Protocol, Src: X.X.X.X (X.X.X.X), Dst: 192.168.2.11 (192.168.2.11) User Datagram Protocol, Src Port: 23036 (23036), Dst Port: 5060 (5060) Session Initiation Protocol Status-Line: SIP/2.0 200 OK Message Header Via: SIP/2.0/UDP 10.1.201.69;received=X.X.X.X;rport=5060;branch=z9hG4bK0a01c9450000002743ff5f3c0000607b0000002e Record-Route: <sip:*02139435245@X.X.X.X:5060;ftag=580759013014;lr=on> From: "Jack Wei"<sip:jack.wei@X.X.X.X:5060>;tag=580759013014 To: <sip:*0@X.X.X.X:5060>;tag=as45d54422 Call-ID: 4AB73688-77E6-43D1-83FC-785BBE46BC4F(a)10.1.201.69 CSeq: 3 BYE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:*02139435245@192.168.2.1:5060> Content-Length: 0 P-hint: NAT traversal

1 0

[Serusers] serweb
by Ofer Zaarur 25 Feb '06

25 Feb '06

Hi, I'm Newbie with ser. I have ser server works fine on my local network. I tried to setup serweb without success serweb9.4. I'm getting ---------------------------------------- Not Found The requested URL /serweb/html/admin/index.php was not found on this server. ________________________________________________________________________ Apache/2.0.54 (Fedora) Server at 192.168.1.45 Port 80 fedoratest page all the time. --------------------------------------- I follow the installation but still doesn't work. is there any guide to help me setup the serweb? is someone has any idea? Thanks, Ofer P.S. i'm using php 5.0.4 mysql 4.1.11 attache are the modified files:

5 13

[Users] Modification of Contact field in SIP header
by Bastian Schern 25 Feb '06

25 Feb '06

Hi everybody, is it possible to modify the Contact field inside the SIP header? E. g. We receive a "Moved Temporarily" (302) message in OpenSER and want to change the Contact field with a different value. Regards Bastian ____________ Virus checked by G DATA AntiVirusKit Version: AVK 16.5736 from 24.02.2006 Virus news: www.antiviruslab.com

2 1

[Serusers] More info: lookup(): '' Not found in usrloc
by Kim Culhan 25 Feb '06

25 Feb '06

The ser.cfg in use is: features-callfwd.cfg 15 2005-09-15 17:37:10Z The portion of the config file with this logic appears to be: lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found - lookup_location"); break; }; route(1); } regards -kim -- w8hdkim(a)gmail.com

1 0

[Serusers] ser and firewall on same server
by Corey 25 Feb '06

25 Feb '06

Dual homed server, operating as a vpn server, has one public and one private ip. Remote client pc's ( scattered around the internet ) connect to server's private ip (via openvpn) tunnel, server NATs traffic originating from private/vpn interface through server's public ip off to the internet at large; all this works fine: pc's route all traffic through vpn. Now, if this server is also running an instance of ser, via which the client pc's register - therefore, the NAT'ing takes place on the server _itself_, rather than on some intermediary server/router _between_ the pc's and ser server - is it still necessary to run mediaproxy or nathelper? I'm still cutting my teeth on voip in general, and ser in particular - however our environment seems quite different to what I have seen and read in all the docs and material regarding sip/ser and the NAT issues involved -- all of these seem to assume that the NAT'ing takes place on external/3rd-party servers/routers rather than on the ser server itself. As such, I'm having all kinds of fun trying to figure out what the heck I'm doing - whether I'm on the right track, and am merely suffering from some small misconfig somewhere that is preventing all this to work, or whether this scenerio is so outlandish that I'm unlikely to ever succeed period. So what I'm hoping to get, is some indication from you experts on whether this environment I've tried to conceptualize above is possible or not; and any tips or suggestions you may have. I'll be happy to provide further clarification on anything that I haven't described clearly. Many thanks! Cheers, Corey

1 0

[Serusers] local ACK found -> dropping it!
by Corey 25 Feb '06

25 Feb '06

Having some troubles with a new voip gateway service we're using; hoping someone could provide any advice. To sum up: Call is initiated from pots line; gets routed to voip gateway provider; gateway forwards invite to our ser server; one ring then busy signal is heard on originating/requesting pots line; the destined softphone UA, registered via our ser server, never gets call. As far as I can fathom, ser recieves an INVITE forwarded/routed from the gateway, then ser requests auth/cred, then gateway responds with an ACK and at that point everything just stops. Assistance is much appreciated, as I'm unable to determine what may be causing the issue. 208.xxx.xxx.xx = 3rd-party/commercial voip gateway server 206.xx.xx.x = our ser proxy server 1234567890 = originating pots line 8889997777 = softphone UA number ( aliased to sip uri 'blah(a)206.xx.xx.x' ) ngrep: # U 208.xxx.xxx.xx:5060 -> 206.xx.xx.x:5060 INVITE sip:8889997777@206.xx.xx.x SIP/2.0..Via: SIP/2.0/UDP 208.xxx.xxx.xx:5060;branch=z9hG4bK40f0830c;rport..From: "John Doe" <sip:1234567890@208.xxx.xxx.xx>;tag=as5f35b551..To: <s ip:8889997777@206.xx.xx.x>..Contact: <sip:1234567890@208.xxx.xxx.xx>..Call-ID: 02c132b659e84c2321debbf226051e3d@208.xxx.xxx.xx..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70 ..Date: Fri, 24 Feb 2006 22:28:05 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY..Content-Type: application/sdp..Content-Length: 397....v=0..o=root 32669 32669 IN IP4 208.xxx.xxx.xx..s=session..c=IN IP4 208.xxx.xxx.xx..t=0 0..m=audio 10784 RTP/AVP 0 18 8 3 111 4 97 101..a=rtpmap:0 PCMU/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:8 PCMA/80 00..a=rtpmap:3 GSM/8000..a=rtpmap:111 G726-32/8000..a=rtpmap:4 G723/8000..a=rtpmap:97 iLBC/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -.. # U 206.xx.xx.x:5060 -> 208.xxx.xxx.xx:5060 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 208.xxx.xxx.xx:5060;branch=z9hG4bK40f0830c;rport=5060..From: "John Doe" <sip:1234567890@208.xxx.xxx.xx>;tag=as5f35b551..To : <sip:8889997777@206.xx.xx.x>;tag=2d38da4c38ea547b795405b32554c3bb.f3cb..Call-ID: 02c132b659e84c2321debbf226051e3d@208.xxx.xxx.xx..CSeq: 102 INVITE..Proxy-Authenticate: Digest realm="208.1 39.204.245", nonce="43fe339b33d7c58a0d89e4711838584d"..Server: Sip EXpress router (0.9.4 (i386/linux))..Content-Length: 0..Warning: 392 206.xx.xx.x:5060 "Noisy feedback tells: pid=2 974 req_src_ip=208.xxx.xxx.xx req_src_port=5060 in_uri=sip:8889997777@206.xx.xx.x out_uri=sip:8889997777@206.xx.xx.x via_cnt==1".... # U 208.xxx.xxx.xx:5060 -> 206.xx.xx.x:5060 ACK sip:8889997777@206.xx.xx.x SIP/2.0..Via: SIP/2.0/UDP 208.xxx.xxx.xx:5060;branch=z9hG4bK40f0830c;rport..From: "John Doe" <sip:1234567890@208.xxx.xxx.xx>;tag=as5f35b551..To: <sip: 8889997777(a)206.xx.xx.x>;tag=2d38da4c38ea54795405554c3bb...Contact: <sip:1234567890@208.xxx.xxx.xx>..Call-ID: 02c132b659e84c2321debbf226051e3d@208.xxx.xxx.xx..CSeq: 102 ACK..User-A gent: Asterisk PBX..Max-Forwards: 70..Content-Length: 0.... ... end of transmission; no further exchange after gateway ACK; originating pots line hears a single ring, then fast busy signal; softphone UA never recieves INVITE or anything ser debug: IP Request: 0(2974) method: <INVITE> 0(2974) uri: <sip:8889997777@206.xx.xx.x> 0(2974) version: <SIP/2.0> 0(2974) parse_headers: flags=1 0(2974) Found param type 232, <branch> = <z9hG4bK40f0830c>; state=6 0(2974) Found param type 235, <rport> = <n/a>; state=17 0(2974) end of header reached, state=5 0(2974) parse_headers: Via found, flags=1 0(2974) parse_headers: this is the first via 0(2974) After parse_msg... 0(2974) preparing to run routing scripts... 0(2974) parse_headers: flags=128 0(2974) end of header reached, state=9 0(2974) DEBUG: get_hdr_field: <To> [31]; uri=[sip:8889997777@206.xx.xx.x] 0(2974) DEBUG: to body [<sip:8889997777@206.xx.xx.x> ] 0(2974) get_hdr_field: cseq <CSeq>: <102> <INVITE> 0(2974) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2974) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2974) DEBUG: add_param: tag=as5f35b551 0(2974) end of header reached, state=29 0(2974) parse_headers: flags=256 0(2974) DEBUG: get_hdr_body : content_length=397 0(2974) found end of header 0(2974) find_first_route: No Route headers found 0(2974) loose_route: There is no Route HF 0(2974) is_local(): Realm '206.xx.xx.x' is local 0(2974) parse_headers: flags=16384 0(2974) pre_auth(): Credentials with given realm not found 0(2974) build_auth_hf(): 'Proxy-Authenticate: Digest realm="208.xxx.xxx.xx", nonce="43fe35e639b33d7c58a0d89e47df7fe11838584d" ' 0(2974) parse_headers: flags=-1 0(2974) check_via_address(208.xxx.xxx.xx, 208.xxx.xxx.xx, 0) 0(2974) DEBUG:destroy_avp_list: destroying list (nil) 0(2974) receive_msg: cleaning up 0(2974) SIP Request: 0(2974) method: <ACK> 0(2974) uri: <sip:8889997777@206.xx.xx.x> 0(2974) version: <SIP/2.0> 0(2974) parse_headers: flags=1 0(2974) Found param type 232, <branch> = <z9hG4bK40f0830c>; state=6 0(2974) Found param type 235, <rport> = <n/a>; state=17 0(2974) end of header reached, state=5 0(2974) parse_headers: Via found, flags=1 0(2974) parse_headers: this is the first via 0(2974) After parse_msg... 0(2974) preparing to run routing scripts... 0(2974) parse_headers: flags=4 0(2974) DEBUG: add_param: tag=2d38da4c38ea547b795405b32554c3bb.f3cb 0(2974) end of header reached, state=29 0(2974) DEBUG: get_hdr_field: <To> [73]; uri=[sip:8889997777@206.xx.xx.x] 0(2974) DEBUG: to body [<sip:8889997777@206.xx.xx.x>] 0(2974) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 0(2974) DEBUG:destroy_avp_list: destroying list (nil) 0(2974) receive_msg: cleaning up

2 2

[Users] Custom CNAM module
by Brandon Price 24 Feb '06

24 Feb '06

I am interested in a custom openser module. It needs to perform cnam (calling party name) resolution through a third party webserver. The module should perform the following actions on an incoming call where SIP Display info: == " "(nothing). 1.Query our local cnam_custom db (mysql) to determine if the customer being called has a custom cnam for the number calling them 2.Query our local cnam_cache db (mysql) to determine if we have already queried the third party database for the number calling in 3.Query the third party database (https) to determine cnam if neither of the first two queries returned anything 4.Insert the result into the SIP Display info: portion of the INVITE so that the correct name will show up on my users callerid. If 1,2,or 3 return a non null value skip to to 4. I would really love to do this with an external script that grabs the values, inserts them into the database, and AVP's but have been this far been unable to build a working solution. I tried... avp_db_load("$from","cnam/$cnam_scheme"); xlog("cnam value [$avp(s:cnam)]\n"); uac_replace_from("$cnam",""); or something like that to no avail. Please reply with your hourly rate if you will need to write the module, if you can do it with $avp's?, and include total cost for the job. I would prefer to do it with avp's. Thank you.

2 1

R: [Serusers]
by Piccari Gianluca 24 Feb '06

24 Feb '06

Hi all, does anyone know if SER supports Number Portability. Thank you in advance

2 1

[Serusers] serweb question
by frankie chiang 24 Feb '06

24 Feb '06

Dear all, 1. How can I change username prefix 8 to other under user registration ? 2. How can I set default timezone for all users ? Best Regard, __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers]
by Piccari Gianluca 24 Feb '06

24 Feb '06

Hi all, does anyone know if SER supports Number Portability. Thank you in advance -----Messaggio originale----- Da: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]Per conto di gokhan uslu Inviato: venerdì 24 febbraio 2006 16.54 A: serusers(a)lists.iptel.org Oggetto: [Serusers] radius authentication with ip address of a user Hi, I have installed ser and freeradius into same machine. everything is working fine. i have created users at the freeradius. but while creating a user, in the user information i had to use the ip of the SER not the ip of the user gateway(a sip gateway) I mean i want to register the user according to its username , password and ALSO IP. my user`s ip: 83.100.100.23 my SER ip: 212.175.10.36 when i edit user file for freeradius and when i write this: myuser(a)83.100.100.23 for user name it does not work but when i write myuser(a)212.175.10.36 then it can authenticate. but this is unusefull for me at the moment. I must be authenticated according to user`s ip, not according to SER`s ip please help meee thanks a lot _____ Relax. Yahoo! Mail virus <http://us.rd.yahoo.com/mail_us/taglines/virusall/*http://communications.yah…> scanning helps detect nasty viruses!

1 0

[Serusers] radius authentication with ip address of a user
by gokhan uslu 24 Feb '06

24 Feb '06

Hi, I have installed ser and freeradius into same machine. everything is working fine. i have created users at the freeradius. but while creating a user, in the user information i had to use the ip of the SER not the ip of the user gateway(a sip gateway) I mean i want to register the user according to its username , password and ALSO IP. my user`s ip: 83.100.100.23 my SER ip: 212.175.10.36 when i edit user file for freeradius and when i write this: myuser(a)83.100.100.23 for user name it does not work but when i write myuser(a)212.175.10.36 then it can authenticate. but this is unusefull for me at the moment. I must be authenticated according to user`s ip, not according to SER`s ip please help meee thanks a lot --------------------------------- Relax. Yahoo! Mail virus scanning helps detect nasty viruses!

1 0

[Users] BYE retransmits too quick!
by Greg Fausak 24 Feb '06

24 Feb '06

I'm doing an interop test with a well known ITPS, they are kicking back these BYE retransmits, like: SIP MESSAGE 18 216.138.115.30:5061(www.testnode-3.com) -> 209.247.16.2:5060(deneps551.Denver1) UDP Frame 18 17/Feb/06 17:09:36.8142 TimeFromPreviousSipFrame=0.0010 TimeFromStart=8.0692 BYE sip:209.247.16.2:5060 SIP/2.0 Record-Route: <sip:216.138.115.30:5061;ftag=245-192.168.10.1;lr> Record-Route: <sip:216.138.115.27;ftag=245-192.168.10.1;lr> Via: SIP/2.0/UDP 216.138.115.30:5061;branch=z9hG4bK036d.a99621d6.0 Via: SIP/2.0/UDP 216.138.115.27;branch=z9hG4bK036d.1fe4ca41.0 Via: SIP/2.0/UDP 198.41.9.65:5060;branch=z9hG4bK245-192.168.10.1.256 Via: SIP/2.0/UDP 192.168.10.23:5060;branch=z9hG4bK1371-192.168.10.23.1374 From: <sip:+18663072489@216.138.115.30:5061>;tag=245-192.168.10.1 To: "Unknown" <sip:+17205626376@209.247.16.2>;tag=VPSF50603522629634 Call-ID: DEN05020060217230928060500(a)209.244.48.214 CSeq: 101 BYE Max-Forwards: 67 Record-Route: <sip:198.41.9.65:5060> Contact: "8663072489" <sip:+18663072489@198.41.9.65:1074> SIP MESSAGE 19 216.138.115.30:5061(www.testnode-3.com) -> 209.247.16.2:5060(deneps551.Denver1) UDP Frame 19 17/Feb/06 17:09:36.8424 TimeFromPreviousSipFrame=0.0282 TimeFromStart=8.0974 BYE sip:209.247.16.2:5060 SIP/2.0 Record-Route: <sip:216.138.115.30:5061;ftag=245-192.168.10.1;lr> Record-Route: <sip:216.138.115.27;ftag=245-192.168.10.1;lr> Via: SIP/2.0/UDP 216.138.115.30:5061;branch=z9hG4bK036d.a99621d6.0 Via: SIP/2.0/UDP 216.138.115.27;branch=z9hG4bK036d.1fe4ca41.0 Via: SIP/2.0/UDP 198.41.9.65:5060;branch=z9hG4bK245-192.168.10.1.256 Via: SIP/2.0/UDP 192.168.10.23:5060;branch=z9hG4bK1371-192.168.10.23.1374 From: <sip:+18663072489@216.138.115.30:5061>;tag=245-192.168.10.1 To: "Unknown" <sip:+17205626376@209.247.16.2>;tag=VPSF50603522629634 Call-ID: DEN05020060217230928060500(a)209.244.48.214 CSeq: 101 BYE Max-Forwards: 67 Record-Route: <sip:198.41.9.65:5060> Contact: "8663072489" <sip:+18663072489@198.41.9.65:1074> Why is the BYE being sent so quickly? This is an intermediate UAS proxy, and every now and then it spits out an extra BYE like this. Shouldn't this wait 500ms before doing a retry? The rest of the trace is clean. I can forward the whole thing if there is any interest. -g -- Greg Fausak greg(a)thursday.com

3 5

Re: [Users] Openser TLS: problem on port 5061
by Klaus Darilion 24 Feb '06

24 Feb '06

Hi Martin! please always cc the list Martin Petraschek wrote: > I want openser to listen on port 5060 for UDP and TCP, and on port 5061 > for TCP/TLS. How could I achieve this? My current config is as follows: > > listen=udp:10.1.2.9:5060 > listen=tcp:10.1.2.9:5060 > disable_tls = 0 > listen = tls:10.1.2.9:5061 this is fine > tls_port_no=5061 this is not needed > > Is this correct? > > > What is the request URI and destination before sending (try > > xlog("L_ERR","duri=$du, ruri=$ru") before t_relay()) > > With the openser.cfg as shown above the error message I described in my > mail previous mail went away. However, when I try to make a call, I get > the following: > > Feb 24 15:01:48 server2 /usr/local/sbin/openser[4246]: duri=, > ruri=sip:user2@192.168.0.47:3985;transport=tls;line=ojn9itpa > Feb 24 15:01:49 server2 /usr/local/sbin/openser[4223]: child process > 4246 exited by a signal 11 Sorry, i have no idea why it crashes here. Try to increase the debug level (debug=4) and watch the logfile. regards klaus > Feb 24 15:01:49 server2 /usr/local/sbin/openser[4223]: core was generated > Feb 24 15:01:49 server2 /usr/local/sbin/openser[4223]: INFO: terminating > due to SIGCHLD > > If you need more information, please tell me. > > Thanks, > > Martin > > > >

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Re: [Serusers] multi video conferencing
by Giovanni Balasso 24 Feb '06

24 Feb '06

try Eyebeam, non free, but works great with ser: http://www.xten.net/index.php Ciao --- Piergiorgio Venuti <piergiorgio(a)mediaservice.net> ha scritto: > Hi all, > who know any windows program that support multi video conferencing with > ser, Windows Messanger work well with ser but not sopport this feature. > Any idea? > > Tnx > > -- > +----------------------------------------------------------------------+ > | Venuti Piergiorgio Email: piergiorgio(a)mediaservice.net | > | System Manager Tel: +39-011-32.72.100 | > | @ Mediaservice.net S.R.L. Fax: +39-011-32.46.497 | > | Via S.Bernardino 17 Torino | > | Disclaimer: http://@Mediaservice.net/disclaimer | > +----------------------------------------------------------------------+ > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > ___________________________________ > Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB > http://mail.yahoo.it > ___________________________________ Yahoo! Mail: gratis 1GB per i messaggi e allegati da 10MB http://mail.yahoo.it

1 0

[Users] Question about the treatment of incoming messages
by Antoine Fressancourt 24 Feb '06

24 Feb '06

Hello, I have a question about the behaviour of OpenSER while treating incoming messages. This question may be a detail for most of the people here, but I think that it has an impact on the dimensioning of an OpenSER setup, especially if it is used to treat a large number of users / messages. I would like to know if Openser is able to store incoming messages in a queue when all the children launched by the program are occupied, or if the messages are rejected when no threads are available to treat them. For instance, if I launch openser with children=5 and these 5 children threads are occupied when a SIP message arrives, is the message rejected or is it stored in a FIFO queue ? In addition, I would be interested to know which part of the code is relevant to look at to deal with this issue. Thanks in advance for all your remarks and answers about my question. Antoine Fressancourt

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[Users] Few questions about dynamic routing and load management
by Florian Fainelli 24 Feb '06

24 Feb '06

Hello all ! First of all, thank you very much for this great software it is really awesome. For the moment, I am doing static routing with config file rotation in order to have a kind of dynamic routing to voip gateways, but I would like to store my routing conditions (src_ip, forward ...) in a MySQL database, is that currently possible ? I also have another need. Is openser capable of knowing the capacity of a remote gateway it fowards the traffic to ? For instance, I have a machine sending 100+ calls, and I have several voip gateways with 32 channels, then I would like to "fill" them according to my openser incoming calls. I am not sure I am clear, but I can re-explain that if needed. Finally, is there a management tool that can show at any time the amount of calls being reached trough openser ? Thank you very much in advance for your answers and keep on the good work ! -- Cordialement, Florian Fainelli --------------------------------------------- 5, rue Charles Fourier Chambre 1511 91011 Evry http://www.alphacore.net (+33) 01 60 76 64 86 (+33) 06 09 02 64 95 --------------------------------------------- Institut National des Télécomnications http://www.int-evry.fr/telecomint ---------------------------------------------

2 1

[Users] Error when trying to lookup locations in a non local database
by Dioris Moreno 24 Feb '06

24 Feb '06

I have three sip proxies configured with OpenSER and MediaProxy for NAT traversal support, I'm going to call them Proxy A, B and C. These proxies work together as a farm; the Proxies A and C points to the MySql database installed in Proxy B. Proxy B serves as location server an also as proxy. Everything works fine, the REGISTER requests are validated against the database (installed in Proxy B) by any proxy without any problem. Also, the location records are inserted without any problem. However, I noticed when Proxies A or C receive an INVITE request, they respond 404 - User Not Found. I made a lot of debug, captures and anything else, and I found the lookup function is only getting locations for user agents registered by the Proxy that is making the query; in other words, if the user X registers through Proxy A, then an INVITE request received by Proxy A to user X is routed successfully. I don't know if I'm missing something. I'm using db_mode 1 for usrloc to store locations directly in the database. Below is one of my configuration scripts, the three are practically the same, they are different on the listen IP addresses and in the record route for nated clients. I'll appreciate any help. Thank you # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=AAA.BBB.CCC.DDD port=5060 children=4 fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:password@dbs.sip.mydomain.com/openser" #Aliases alias="sip.mydomain.com" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/alias_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc|auth_db|domain|uri_db|alias_db", "db_url", "mysql://openser:password@dbs.sip.mydomain.com/openser") modparam("usrloc", "use_domain", 0) modparam("usrloc", "db_mode", 1) modparam("usrloc", "timer_interval", 60) modparam("registrar", "use_domain", 0) modparam("registrar", "nat_flag", 6) modparam("registrar", "min_expires", 10) modparam("registrar", "max_expires", 80) modparam("registrar", "default_expires", 30) modparam("rr", "enable_full_lr", 1) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-clie nts") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-clie nts") # ------------------------- request routing logic ------------------- # main routing logic route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("AAA.BBB.CCC.DDD:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); exit; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); exit; }; if (method=="CANCEL") { route(1); exit; } else if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; } else if (method=="ACK") { exit; }; if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- xlog("L_INFO", "REGISTER - entering section\n"); if (client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); xlog("L_INFO", "REGISTER - nated client detected and fixed ($tu)\n"); }; if (registered("location")) { xlog("L_INFO", "REGISTER - user already registered ($tu)\n"); }; if (!www_authorize("sip.mydomain.com","subscriber")) { www_challenge("sip.mydomain.com","0"); exit; }; if (!check_to()) { xlog("L_WARN", "REGISTER - unauthorized client ($tu)\n"); sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); xlog("L_INFO", "REGISTER - client authorized ($tu)\n"); if (!save("location")) { xlog("L_WARN", "REGISTER - client location could not be saved ($tu)\n"); sl_reply_error(); }; xlog("L_INFO", "REGISTER - client location saved ($tu)\n"); } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- xlog("L_INFO", "INVITE - entering section\n"); #if (client_nat_test("3")) { if (client_nat_test("7")) { setflag(7); fix_nated_contact(); force_rport(); }; if (!proxy_authorize("sip.mydomain.com","subscriber")) { proxy_challenge("sip.mydomain.com","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); xlog("L_INFO", "INVITE - proxy authorized\n"); if (uri!=myself) { xlog("L_INFO", "INVITE - <$ru> uri is not from here, routing out\n"); route(4); route(1); exit; }; xlog("L_INFO", "INVITE - before aliases lookup <$ru>\n"); alias_db_lookup("dbaliases"); xlog("L_INFO", "INVITE - after aliases lookup <$ru>\n"); if (registered("location")) { xlog("L_INFO", "INVITE - the user is already registered <$ru>\n"); }; if (!lookup("location")) { xlog("L_WARN", "INVITE - not located <$ru>\n"); sl_send_reply("404", "User Not Found"); exit; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { xlog("L_INFO", "NAT Traversal - using media proxy\n"); use_media_proxy(); }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

2 1

[Users] Openser TLS: problem on port 5061
by Martin Petraschek 24 Feb '06

24 Feb '06

Hi all! I finally got two clients (minisip and Snom softphone) to work with openser via TLS. I was using port 5060 for TLS connections, which is non-standard. I wanted to change this behavious and make openser listen for TLS connections on port 5061, which I thought should be possible with the following configuration parameter: listen = tls:10.1.2.9:5061 After configuring the client (minisip) to use port 5061 for TLS, restarting openser and re-registering the client, I get the following error messages in the openser log: Feb 23 16:48:45 server /usr/local/sbin/openser[6011]: ERROR:tm:add_uac: can't fwd to af 2, proto 1 (no corresponding listening socket) Feb 23 16:48:45 server /usr/local/sbin/openser[6011]: ERROR:tm:t_forward_nonack: failure to add branches The ssl handshake worked without problems, which I confirmed by looking at the output of ssldump. Can anyone tell me what these error messages mean? I am using openser-1.0.0-tls. Cheers, Martin

2 1

[Serusers] How to add DNS SRV records for mediaproxy (proxy dispatcher)
by Ramu Yadav 24 Feb '06

24 Feb '06

Hi Folks, Can anybody help me how to add DNS SRV records for mediaproxy and proxy dispatcher load balancing concept. I appreciate your help. -- Ramu Yadav

1 0

[Serusers] SER+NAT
by Barta János 24 Feb '06

24 Feb '06

Hello! I would like to make a SIP proxy, with dual function. The first function will be a normal SIP proxy, while in the sec. function it will be working as an RTP proxy. For example: A user sends a registration reguest to my SIP. The proxy sees, that the domain name is mytest.ser, and the proxy will know, that this domain will be managed by itself. If the user creates a new registration -but in this case with mytest.hu- then the SER will do a NAT, and forwarding the message away, via the external interface (to another proxy). The arch: Client <-> DefaultGW (+SIP and RTP proxy) <-> Another SIP 10.1.1.1 <-> 10.1.1.254 __NAT___10.200.200.254 <-> 10.200.200.1 The default config -with a little modification- is working fine, but it does not rewrite the contact header in the SDP file . (I installed the rtpproxy, and I have set up iptables too.) alias=mytest.ser loadmodule "/usr/lib/ser/modules/nathelper.so" route{ #inserted by klaus if (method=="INVITE") { record_route(); force_rtp_proxy(); forward(10.200.200.1,5060); /* set up reply processing */ t_on_reply("1"); }; forward(10.200.200.1,5060); } #inserted by klaus # all incoming replies for t_onrepli-ed transactions enter here onreply_route[1] { if (status=~"[12][0-9][0-9]") force_rtp_proxy(); } A number of persons will be use it with M$ Messenger, so I'm interested in such a solution, where we do not need to set an outbound proxy in the client. Can somebody send me any example? Thanks, Jani ___________________________________________________________________________ [origo] klikkbank lakossági számlacsomag havi 199 Ft-ért, bankkártya éves díj nélkül! www.klikkbank.hu

1 0

[Serusers] SER + Asterisk + Billing
by Jean-Michel Hiver 24 Feb '06

24 Feb '06

Hi List, I have the following setup now: <wholesale customer> -> SER -> Asterisk -> PSTN Gateway I use SER to handle all the SIP mambo-jumbo for my wholesale customers and my softphones. I use Asterisk to collect CDRs and to LCR routing. Now here is to my question: For my wholesale customers, I do IP-based "trusted host" authentication. Is there a way to set a SIP field (in SER) which will identify them uniquely in Asterisk's Master.csv file? The Asterisk records can look like this: "","anonymous","0262914900","world","""Anonymous""<anonymous>","SIP/anonymous.invalid-08160370","SIP/mygateway-7073","Dial","SIP/16170262914900@mygateway","2006-02-24 13:03:51","2006-02-24 13:04:08","2006-02-24 13:04:22",31,14,"ANSWERED","DOCUMENTATION" Any ideas? Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

1 0

[Serusers] multi video conferencing
by Piergiorgio Venuti 24 Feb '06

24 Feb '06

Hi all, who know any windows program that support multi video conferencing with ser, Windows Messanger work well with ser but not sopport this feature. Any idea? Tnx -- +----------------------------------------------------------------------+ | Venuti Piergiorgio Email: piergiorgio(a)mediaservice.net | | System Manager Tel: +39-011-32.72.100 | | @ Mediaservice.net S.R.L. Fax: +39-011-32.46.497 | | Via S.Bernardino 17 Torino | | Disclaimer: http://@Mediaservice.net/disclaimer | +----------------------------------------------------------------------+

1 0

[Serusers] RTPProxy
by George Lambson 24 Feb '06

24 Feb '06

Hi, Does anyone know if RTPProxy and NATProxy support asymmetric RTP UAs? Thanks, George

2 1

[Serusers] install probs with config
by alex 23 Feb '06

23 Feb '06

i am running fedora core and trying to install the latest version of ser i amm useing xampp till i can get it working properly and finish writig the guid eim writting then i will install everything properly but the errors im getting arnt releted to me have xampp on my box. i used dan austins how to to set up i got as far as the config file edits beofe it went bad. wehn i tryed to start it i got ERROR: bad config file (12 errors) so i ran the debug for it (-c) and here is what i got 0(14242) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.12: cannot open shared object file: No such file or directory 0(14242) parse error (28,13-14): failed to load module 0(14242) parse error (113,11-15): syntax error 0(14242) parse error (113,11-15): bad command 0(14242) parse error (113,16-18): bad command 0(14242) parse error (113,23-27): bad command 0(14242) parse error (113,28-30): bad command 0(14242) parse error (113,31-34): bad command 0(14242) parse error (113,35-41): bad command 0(14242) parse error (113,42-56): bad command 0(14242) parse error (114,4-6): bad command 0(14242) parse error (121,4-5): syntax error 0(14242) parse error (121,4-5): ERROR: bad config file (12 errors) and here is my config file # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==perfecthosting.biz) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==perfecthosting.biz) { if (method=="REGISTER") { Uncomment this if you want to use digest authentication if (!www_authorize("perfecthosting.biz", "subscriber")) { www_challenge("perfecthosting.biz", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==perfecthosting.biz) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } i removed the dots from the modula locations because they were causeing errors it used to sat 40 something errors. im lost as far as syntax or the mysql so if you have any insight about this it would be good if you could share thanks, Alex Wood

3 2

[Serusers] OpenSER ACC error
by Edson 23 Feb '06

23 Feb '06

Sorry if this is the wrong list, but.... The scenario: We're migrating from an SER 0.9.2 working installation to an OpenSER 1.0.0. The problem.: after tweaking directories names and substituting 'break' by 'exit', the output of 'openser -c /etc/openser/openser.cfg' reports problems on assigning 'service_type', 'radius_flag', 'radius_config' and 'radius_missed_flag' ACC module variables. The config...: below config relevant fragments... loadmodule "/usr/lib/openser/modules/mysql.so" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/auth.so" loadmodule "/usr/lib/openser/modules/textops.so" loadmodule "/usr/lib/openser/modules/lcr.so" loadmodule "/usr/lib/openser/modules/uri.so" loadmodule "/usr/lib/openser/modules/uri_db.so" loadmodule "/usr/lib/openser/modules/auth_radius.so" loadmodule "/usr/lib/openser/modules/uri_radius.so" loadmodule "/usr/lib/openser/modules/avpops.so" loadmodule "/usr/lib/openser/modules/acc.so" loadmodule "/usr/lib/openser/modules/domain.so" loadmodule "/usr/lib/openser/modules/permissions.so" . . . modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 1 ) modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_missed_flag", 2 ) . . . The Error: sip:/etc/openser # openser -c openser.cfg 0(5264) set_mod_param_regex: parameter <service_type> not found in module <acc> 0(5264) parse error (44,42-43): Can't set module parameter 0(5264) set_mod_param_regex: parameter <radius_flag> not found in module <acc> 0(5264) parse error (45,43-44): Can't set module parameter 0(5264) set_mod_param_regex: parameter <radius_config> not found in module <acc> 0(5264) parse error (46,34-35): Can't set module parameter 0(5264) set_mod_param_regex: parameter <radius_missed_flag> not found in module <acc> 0(5264) parse error (47,36-37): Can't set module parameter ERROR: bad config file (4 errors) What am I missing? Edson.

2 1

[Serusers] Delete header fields
by Xavier TRENTIN 23 Feb '06

23 Feb '06

Hi all, is there a way to delete a header field from the header part of a sip request? Thanks in advance. Xavier.

2 1

[Users] path and received in location
by Helge Waastad 23 Feb '06

23 Feb '06

Hi, I've got a question regarding the handling of location information. I came up in a situation where I had my dispatcher layer adding path to the register before it was sent to the registrar. However, if the register then again is replicated to another server, and the next server is adding a received to the usrloc entry, how should this server forward the INVITES...? To the "path" or to the "received"? PS, I'm not saying this is how it should be done, I'm just curious about the implications :-) br hw Mvh, Helge Waastad Senior Engineer Smartnet tlf: 67830017

3 2

[Serusers] Error checking on a forwarded route
by sip 23 Feb '06

23 Feb '06

Is there any way to catch errors on a forwarded route? I have a few SIP codes that are forwarded to other machines using rewritehostport. Once in a while, those machines will be down or doing something odd, and I'll get a 404 response from them. Is there any way for SER to catch/handle those returns? I've been playing around with the onreply_route, but I'm getting weird errors, so I thought I'd see if I was headed in the right direction before I go through the hassle of figuring out why I'm getting things in the log like: ERROR: w_t_relay_to: unsupported mode: 2 and such. N.

1 0

[Serusers] Re: [Serhelp] Problem with SER accounting, Please help
by mkumar＠mantragroup.com 23 Feb '06

23 Feb '06

Hi, Thanks for the reply. My problem is solved by changing value of db_localtime in acc module and recompiling it. Manoj. Quoting Jan Janak <jan(a)iptel.org>: > Hello, > > SER does know the timezone of the caller, thus the timestamp is either in > the timezone of the machine running SER or timezone of the MySQL. > > Configuring MySQL timezone can be a bit tricky, please consult the MySQL > documentation. > > Jan. > > mkumar(a)mantragroup.com wrote: >> Hi All, >> >> I was able to install SER and able to get all the call details to >> the Mysql DB. >> But my problem is with the time of timestamp recorded, it is not recorded >> according to time on Server where my DB is located. The time recorded is >> according to timezone of the caller and so because of this when I >> use NOW() of >> Mysql and get the calls I miss the calls dialled from timezones >> ahead of NOW() >> time on Mysql. I think while inserting call details in to DB the >> acc.so module >> is using some other thing other than using NOW(). I do not know >> where to change >> this, please tell me where can I modify that query and get a common timezone >> for all the users so that billing will not be a problem for me. >> >> Thanks, >> Manoj. >> >> _______________________________________________ >> Serhelp mailing list >> serhelp(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serhelp >> > >

1 0

[Users] SipX/SER
by Chris Roberts 23 Feb '06

23 Feb '06

Heya, I'm looking at implementing a few SipX boxes. These boxes will be registering phones that will be behind firewalls. Currently the SipX support for NAT is rather limited, therefore I'm looking at supplementing SipX with SER on the front-end, basically using OpenSER as an SBC to sort out the NAT for the clients. Does anyone have any configurations that are similar to this configuration before I start from scratch, or even a good starting point? Cheers, Chris. -- No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.1.375 / Virus Database: 268.0.0/266 - Release Date: 21/02/2006

3 4

[Users] build optimalization
by Helge Waastad 23 Feb '06

23 Feb '06

Hi, I'm not really a hardware guy, but just wondering about the tuning parameters for OpenSER. I see -mtune=athlon on my PIII and PIV's Is it worth setting CPU=pentium3/pentium4 when compiling openSER? (Makefile.defs) br hw -- Helge Waastad Senior Konsulent Systemavdelingen Smartnet

4 4

[Users] warning in the log
by unplug 23 Feb '06

23 Feb '06

I have the following messages in the log which I don't know what they mean. After tracing the configuration file, the setting remove_hf("From") will cause one of the message in the log. Do the messages below will affect the system? It seems that there is some mis-configure in the file. How can I correct it if my guess is right? Feb 23 11:37:55 dev01 /usr/local/sbin/openser[31580]: INFO:textops:hname_fixup: using hdr type (30) instead of <Remote-Party-ID> Feb 23 11:37:55 dev01 /usr/local/sbin/openser[31580]: INFO:xl_parse_name: using hdr type (4) instead of <from> Feb 23 11:37:55 dev01 /usr/local/sbin/openser[31580]: INFO:textops:hname_fixup: using hdr type (4) instead of <From>

2 1

[Serusers] Latest presence based ser
by sayan.chowdhury＠wipro.com 23 Feb '06

23 Feb '06

Hi , I am a new SER user. I want to run the ser with the pa and the rls modules. I have tried downloading some of the release versions and installing them but I am getting multiple issues while running them (Invalid table version being the main one) Can anyone direct me to the latest more or less stable version of ser where the pa and the rls modules are known to work ? Regards , Sayan P.S. I am behind a firewall which prevents me from using CVS to download. So web link will be good. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[Serusers] SerWeb help
by Marco Passarinho 23 Feb '06

23 Feb '06

Everytime I try to authenticate in http://mydomain.com/serweb , It appears the following message: DB Error: no such field Bad username or password How can I find out whats going wrong here? Where can I find de logs comming from errors on the queries to the postgres database? Thanks

2 3

[Serusers] Problem with SER accounting, Please help
by mkumar＠mantragroup.com 23 Feb '06

23 Feb '06

Hi All, I was able to install SER and able to get all the call details to the Mysql DB. But my problem is with the time of timestamp recorded, it is not recorded according to time on Server where my DB is located. The time recorded is according to timezone of the caller and so because of this when I use NOW() of Mysql and get the calls I miss the calls dialled from timezones ahead of NOW() time on Mysql. I think while inserting call details in to DB the acc.so module is using some other thing other than using NOW(). Please tell me where can I modify that query and get a common timezone for all the users so that billing will not be a problem for me. Any help will be sincerely appreciated. Thanks, Manoj.

1 0

[Users] Error when trying to lookup locations in a non local database
by Dioris Moreno 23 Feb '06

23 Feb '06

I have three sip proxies configured with OpenSER and MediaProxy for NAT traversal support, I'm going to call them Proxy A, B and C. These proxies work together as a farm; the Proxies A and C points to the MySql database installed in Proxy B. Proxy B serves as location server an also as proxy. Everything works fine, the REGISTER requests are validated against the database (installed in Proxy B) by any proxy without any problem. Also, the location records are inserted without any problem. However, I noticed when Proxies A or C receive an INVITE request, they respond 404 - User Not Found. I made a lot of debug, captures and anything else, and I found the lookup function is only getting locations for user agents registered by the Proxy that is making the query; in other words, if the user X registers through Proxy A, then an INVITE request received by Proxy A to user X is routed successfully. I don't know if I'm missing something. I'm using db_mode 1 for usrloc to store locations directly in the database. Below is one of my configuration scripts, the three are practically the same, they are different on the listen IP addresses and in the record route for nated clients. I'll appreciate any help. Thank you # ----------- global configuration parameters ------------------------ debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) # Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes # check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=AAA.BBB.CCC.DDD port=5060 children=4 fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:password@dbs.sip.mydomain.com/openser" #Aliases alias="sip.mydomain.com" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/alias_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" # ----------------- setting module-specific parameters --------------- modparam("usrloc|auth_db|domain|uri_db|alias_db", "db_url", "mysql://openser:password@dbs.sip.mydomain.com/openser") modparam("usrloc", "use_domain", 0) modparam("usrloc", "db_mode", 1) modparam("usrloc", "timer_interval", 60) modparam("registrar", "use_domain", 0) modparam("registrar", "nat_flag", 6) modparam("registrar", "min_expires", 10) modparam("registrar", "max_expires", 80) modparam("registrar", "default_expires", 30) modparam("rr", "enable_full_lr", 1) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) #modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","mediaproxy_socket", "/var/run/proxydispatcher.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/openser/sip-clie nts") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/openser/rtp-clie nts") # ------------------------- request routing logic ------------------- # main routing logic route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("AAA.BBB.CCC.DDD:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); exit; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); exit; }; if (method=="CANCEL") { route(1); exit; } else if (method=="INVITE") { route(3); exit; } else if (method=="REGISTER") { route(2); exit; } else if (method=="ACK") { exit; }; if (uri!=myself) { route(4); route(1); exit; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- xlog("L_INFO", "REGISTER - entering section\n"); #if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { if (client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); xlog("L_INFO", "REGISTER - nated client detected and fixed ($tu)\n"); }; if (registered("location")) { xlog("L_INFO", "REGISTER - user already registered ($tu)\n"); }; if (!www_authorize("sip.mydomain.com","subscriber")) { www_challenge("sip.mydomain.com","0"); exit; }; if (!check_to()) { xlog("L_WARN", "REGISTER - unauthorized client ($tu)\n"); sl_send_reply("401", "Unauthorized"); exit; }; consume_credentials(); xlog("L_INFO", "REGISTER - client authorized ($tu)\n"); if (!save("location")) { xlog("L_WARN", "REGISTER - client location could not be saved ($tu)\n"); sl_reply_error(); }; xlog("L_INFO", "REGISTER - client location saved ($tu)\n"); } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- xlog("L_INFO", "INVITE - entering section\n"); #if (client_nat_test("3")) { if (client_nat_test("7")) { setflag(7); fix_nated_contact(); force_rport(); }; if (!proxy_authorize("sip.mydomain.com","subscriber")) { proxy_challenge("sip.mydomain.com","0"); exit; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); exit; }; consume_credentials(); xlog("L_INFO", "INVITE - proxy authorized\n"); if (uri!=myself) { xlog("L_INFO", "INVITE - <$ru> uri is not from here, routing out\n"); route(4); route(1); exit; }; xlog("L_INFO", "INVITE - before aliases lookup <$ru>\n"); alias_db_lookup("dbaliases"); xlog("L_INFO", "INVITE - after aliases lookup <$ru>\n"); if (registered("location")) { xlog("L_INFO", "INVITE - the user is already registered <$ru>\n"); }; if (!lookup("location")) { xlog("L_WARN", "INVITE - not located <$ru>\n"); sl_send_reply("404", "User Not Found"); exit; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { xlog("L_INFO", "NAT Traversal - using media proxy\n"); use_media_proxy(); }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

1 0

[Users] multiple rtp onsame port
by Jayesh Nambiar 23 Feb '06

23 Feb '06

Hi all,I am facing a strange problem while using openser. I am running openser 1.0.0 and while calling, multiple call gets connected on the same port. I am using Audiocodes media gatewaywhich sends calls to Openser and openser then t_relays the call to pstn gateway. When there are simultaneously 8-10 calls going on, there are more than 1 call getting connected on the same port.It is like a cross connection in which you can hear a other phone ring.I get the following log on the Audiocodes gateway:2d:5h:1m:54s Multiple RTP detected on Port 6010. Switch to Destination Stream IP: 0xd8a8a262, Source Port: 34784. Left Stream IP: 0xd8a8a262, Source Port: 34846 [File:TCPIPHal.cpp Line:1181] [CID:0]2d:5h:1m:54s Multiple RTP detected on Port 6010. Active Stream IP: 0xd8a8a262, Source Port: 34784. Rejected Stream IP: 0xd8a8a262, Source Port: 34846 [Code:7003 File:TCPIPHal.cpp Line:1237] [CID:0] s() My openser conf file is as shown: debug=3 fork=yes log_stderror=no dns=no rev_dns=no listen=202.XX.61.XX port=5060 children=4 fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- modparam("rr", "enable_full_lr", 1) modparam("permissions|acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "report_ack", 0) modparam("acc", "db_extra", "src= $si") # ------------------------- request routing logic ------------------- route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; setflag(1); if (method!="REGISTER") { record_route(); }; if (loose_route()) { route(1); return; }; if(uri!=myself) { route(1); } if (uri==myself) { if(method=="INVITE" && allow_trusted()) { strip(2); prefix("36130"); rewritehost("216.XXX.162.XXX"); route(1); return; } else if(!allow_trusted()) { sl_send_reply("403", "Forbidden"); }; }; if (method=="CANCEL") { route(1); return; }; } route[1] { if(!t_relay()) { sl_reply_error(); }; } Can someone please try and help me on this. thanks in advance, jayesh... --------------------------------- Jiyo cricket on Yahoo! India cricket Yahoo! Messenger Mobile Stay in touch with your buddies all the time.

2 2

[Serusers] installing probs
by alex 22 Feb '06

22 Feb '06

i am running fedora core and trying to install the latest version of ser i amm useing xampp till i can get it working properly and finish writig the guid eim writting then i will install everything properly but the errors im getting arnt releted to me have xampp on my box. i used dan austins how to to set up i got as far as the config file edits beofe it went bad. wehn i tryed to start it i got ERROR: bad config file (12 errors) so i ran the debug for it (-c) and here is what i got 0(14242) ERROR: load_module: could not open module </usr/local/lib/ser/modules/mysql.so>: libmysqlclient.so.12: cannot open shared object file: No such file or directory 0(14242) parse error (28,13-14): failed to load module 0(14242) parse error (113,11-15): syntax error 0(14242) parse error (113,11-15): bad command 0(14242) parse error (113,16-18): bad command 0(14242) parse error (113,23-27): bad command 0(14242) parse error (113,28-30): bad command 0(14242) parse error (113,31-34): bad command 0(14242) parse error (113,35-41): bad command 0(14242) parse error (113,42-56): bad command 0(14242) parse error (114,4-6): bad command 0(14242) parse error (121,4-5): syntax error 0(14242) parse error (121,4-5): ERROR: bad config file (12 errors) and here is my config file # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==perfecthosting.biz) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==perfecthosting.biz) { if (method=="REGISTER") { Uncomment this if you want to use digest authentication if (!www_authorize("perfecthosting.biz", "subscriber")) { www_challenge("perfecthosting.biz", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==perfecthosting.biz) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } i removed the dots from the modula locations because they were causeing errors it used to sat 40 something errors. im lost as far as syntax or the mysql so if you have any insight about this it would be good if you could share thanks, Alex Wood

1 0

[Serusers] Number of the call
by Anderson Alves Albuquerque 22 Feb '06

22 Feb '06

How can I control the number of calling in SER? In GnuGK 2.2.X has this option.

1 0

[Users] NEW FEATURE: cacheless usrloc
by Bogdan-Andrei Iancu 22 Feb '06

22 Feb '06

Hi everybody, USRLOC module can now operate in a DB mode : DB_ONLY. This is a cacheless mode - no memory cache is kept, all operation being directly done into DB. This mode allows DB sharing between multiple proxies without the need of additional replication mechanism. Drawbacks: - some performance penalties due intensive DB usage - location watcher disabled (cannot be bind to a record into mem) => PA cannot be used - statistics do not work since events cannot be properly been traced without a mem cache Still to be done: - the get_all_ucontacts function (used by nathelper/mediaproxy for NAT pinging) use raw_query in order to increase performance by filtering the loaded contacts from DB. For more info, see: http://openser.org/docs/modules/1.1.x/usrloc.html#AEN252 regards, bogdan

1 0

[Serusers] Forwarding - why append_branch() ?
by benaudi＠gmx.de 22 Feb '06

22 Feb '06

Hello, perhaps somebody can explain for me why first example works without append_branch() and second example NEED append_branch() to work? In 1. example i rewrite Request-URI in main-route-block. SER calls the new URI successfully. If i make append_branch() here, SER craches. In 2. example i rewrite Request-URI in failure-block. If i do not make append_branch() here, SER does not call the new URI. With append_branch() everything is all right, SER calls successfully the new URI. 1. example ------------------------- modparam("tm", "fr_inv_timer", 10) # set fr_inv_timer to 10 seconds ... route{ ... if (lookup("location")) { ... route(1); break; } else if (method == "INVITE" ) { revert_uri(); avp_db_load("$ruri","s:redirect_number01"); t_on_failure("2"); attr2uri("redirect_number01"); t_relay(); break; } } ------------------------- 2. example (same ser.cfg) ------------------------- failure_route[1] # entered when fr_inv_timer hits, 408 Timeout { revert_uri(); avp_db_load("$ruri","s:redirect_number01"); t_on_failure("2"); attr2uri("redirect_number01"); append_branch(); # why necessary? t_relay(); break; } } ------------------------- Alex -- Lust, ein paar Euro nebenbei zu verdienen? Ohne Kosten, ohne Risiko! Satte Provisionen f�r GMX Partner: http://www.gmx.net/de/go/partner

1 0

[Users] receiving msgs with invalid port numbers
by Papadopoulos Georgios 22 Feb '06

22 Feb '06

Hello, Sometimes Openser receives REGISTER messages where the port in the Via and in the Contact is invalid (some number greater than 65536). Does anyone know what causes this? Is it just a bug at the user agent side that I cannot do anything about? thank you for any help George 3(24398) ERROR: parse_via: invalid port number <101160> 3(24398) ERROR: parse_via on: <SIP/2.0/UDP 62.1.33.62:101160;branch=z9hG4bKYw5qCaB5TTD1PAvN Max-Forwards: 70 User-Agent: IP PHONE 3 V1.49.002 CFG0 From: "psalidas" <sip:psalidas@i-call.gr>;tag=He0iTAjdDXQBQh0M To: "psalidas" <sip:psalidas@i-call.gr> Call-ID: MCk3KLwds4XOTvqi(a)62.1.33.621 CSeq: 8692 REGISTER Contact: <sip:psalidas@62.1.33.62:101160> Expires: 60 Content-Length: 0 > 3(24398) ERROR: parse_via parse error, parsed so far:<SIP/2.0/UDP 62.1.33.62:101160;branch=z9hG4bKYw5qCaB5TTD1PAvN > 3(24398) ERROR: get_hdr_field: bad via 3(24398) ERROR: parse_msg: message=<REGISTER sip:i-call.gr SIP/2.0 Via: SIP/2.0/UDP 62.1.33.62:101160;branch=z9hG4bKYw5qCaB5TTD1PAvN Max-Forwards: 70 User-Agent: IP PHONE 3 V1.49.002 CFG0 From: "psalidas" <sip:psalidas@i-call.gr>;tag=He0iTAjdDXQBQh0M To: "psalidas" <sip:psalidas@i-call.gr> Call-ID: MCk3KLwds4XOTvqi(a)62.1.33.621 CSeq: 8692 REGISTER Contact: <sip:psalidas@62.1.33.62:101160> Expires: 60 Content-Length: 0 Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

RE: [Serusers] Does SER need a public IP interface?
by George Lambson 22 Feb '06

22 Feb '06

I think mediaproxy would be a better choice. It seemed to be easier to configure for me, and it does load balancing if you want it. Since your server is handling RTP and SIP when your are going through a NAT it can give you the capability to handle more than 50 calls at a time. Also, I am not sure if rtpproxy rewrites the SDP information. That would be why your phones are sending back to the internal port instead of the external one. >>> "Bart J. Smit" <bart(a)smits.co.uk> 2/18/2006 9:45:00 AM >>> No, rtpproxy. Is there a difference in NAT support? Bart... -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of George Lambson Sent: 18 February 2006 15:54 To: serusers(a)lists.iptel.org Subject: Re: [Serusers] Does SER need a public IP interface? Are you using mediaproxy? >>> "Bart J. Smit" <bart(a)smits.co.uk> 2/18/2006 7:16 AM >>> I have based my attached SER config file on the very informative 'getting started 5' instructions/sample from http://www.onsip.org/. I have SER on a DMZ with Asterisk on the same box, listening on 5065. I can register two internal phones with SER and talk. I can register a phone to the Asterisk and talk to the rest of the Asterisk infrastructure through IAX2. I entered a log statement for my forward to asterisk and I can see that it is being executed. What I can't get to work is external phones. When I capture the RTP packets with ethereal, I can see that the ones from SER to the phones are properly addressed to the public IP of my NAT-ed phone. However the return packets are addressed to the private IP of SER and thus never make it back. Is this double NAT configuration supported? It looks that way on the getting started diagrams (both SER and Internet phones are behind NAT routers). I understand that the NAT traversal mechanism in SER is there to help clients overcome their NAT. Is there a similar mechanism for the SER server itself, or does SER only work if it has a direct public interface? How do I convince my client phones to send their packets back to the public IP of SER, rather than the private IP? What ser.cfg line determines the source IP that SER gives out to the RTP stream? It would be elegant if I can make this dependent on the source IP of the phones (internal or external) but I really only need it to work for external clients, as internal ones can simply register with Asterisk. Any hints would be most appreciated, Thanks, Bart... _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] See you later at the SER- Pub today at 18!
by Greger V. Teigre 22 Feb '06

22 Feb '06

I'm about to head over to Beer Palace, Aker Brygge. See you guys there around 6pm! g-)

1 0

[Serusers] SER Dial plan
by Rakib Kanak 22 Feb '06

22 Feb '06

Hi, This is Rakib. I install SER on my Fedora core 2 server. Now i want to know how can i configure to dial on mobile (for Bangladesh e.g 88-0-152377618) and pstn. Greetings Rakib

2 2

[Serusers] Problem using SER with NAT
by Muhammad Asif Fayyaz 22 Feb '06

22 Feb '06

can anybody please help me. This is urgent. There is a continuous problem in using SER. SER is not forwarding the packet correctly to client behind NAT. Please have a look at this problem. My network scenerio is: 192.168.0.2 +----------+ | ClientA | +----------+ | | | eth1 eth0 | 192.168.0.1 | 172.25.25.83 +-------------|-------------+ | NAT1 | +-------------|-------------+ | | | 172.25.25.84 | +---------------------------+ |SER Proxy\Registrar, | | STUN Server | +---------------------------+ |172.25.25.85 | | | | 172.25.25.88 +----------+ | ClientC | +----------+ 172.25.25.X ---- LAN (External Network for client A and C) 192.168.0.X ---- Private Network 1 (Internal Network for Client A) ClientA and ClientC are sip clients with STUN functionality embedded. NAT1 is internal LAN NAT. I mean for testing purpose, i have used two machines with two network cards. -------------------------------------------------------- Here is the problem: Both clientA and clientB get registered successfully with Registrar. ClientA get registered with NAT1 address 172.25.25.83. When ClientA starts sip session by sending an invite to clientC, all works fine. But, when ClientC starts the session by inviting clientA, Proxy tries to forward the message invite directly to clientA. Although, clientA has already registered with the registrar and has indicated its contact address as of NAT1. My question is why proxy is not taking the contact address and sending invite there. below is the sequence of messages that I captured using ehtereal on Linux Plateform. source=172.25.25.88:5060 to=172.25.25.84:5060 REGISTER sip:172.25.25.84:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport;branch=z9hG4bK1507969566 From: <sip:c@172.25.25.88:5060>;tag=1981102178 To: <sip:c@172.25.25.88:5060> Call-ID: 1003517860 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 1 REGISTER Contact: <sip:c%94@172.25.25.88:5060> Max-Forwards: 5 User-Agent: eXosip/0.1 Expires: 3600 Supported: 100rel Supported: path Content-Length: 0 source=172.25.25.84:5060 to=172.25.25.88:5060 IP/2.0 200 OK Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1507969566 From: <sip:c@172.25.25.88:5060>;tag=1981102178 To: <sip:c@172.25.25.88:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.0a43 Call-ID: 1003517860 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 1 REGISTER Contact: <sip:c%94@172.25.25.88:5060>;expires=3600 Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 172.25.25.84:5060 "Noisy feedback tells: pid=21679 req_src_ip= 172.25.25.88 req_src_port=5060 in_uri=sip:172.25.25.84:5060 out_uri=sip: 172.25.25.84:5060 via_cnt==1" source=172.25.25.83:1025 to=172.25.25.84:5060 REGISTER sip:172.25.25.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5060;rport;branch=z9hG4bK600315716 From: <sip:a@192.168.0.2:5060>;tag=658770091 To: <sip:a@192.168.0.2:5060> Call-ID: 2091069424 [!at] 192.168.0.2 (replace the [!at] with a @) CSeq: 1 REGISTER Contact: <sip:a@172.25.25.83:1026> Max-Forwards: 5 User-Agent: eXosip/0.1 Expires: 3600 Supported: 100rel Supported: path Content-Length: 0 source=172.25.25.83:1025 to=172.25.25.84:5060 REGISTER sip:172.25.25.84:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.2:5060;rport;branch=z9hG4bK600315716 From: <sip:a@192.168.0.2:5060>;tag=658770091 To: <sip:a@192.168.0.2:5060> Call-ID: 2091069424 [!at] 192.168.0.2 (replace the [!at] with a @) CSeq: 1 REGISTER Contact: <sip:a@172.25.25.83:1026> Max-Forwards: 5 User-Agent: eXosip/0.1 Expires: 3600 Supported: 100rel Supported: path Content-Length: 0 source=172.25.25.84:5060 to=172.25.25.83:1025 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.2:5060 ;rport=1025;branch=z9hG4bK600315716;received=172.25.25.83 From: <sip:a@192.168.0.2:5060>;tag=658770091 To: <sip:a@192.168.0.2:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b319 Call-ID: 2091069424 [!at] 192.168.0.2 (replace the [!at] with a @) CSeq: 1 REGISTER Contact: <sip:a@172.25.25.83:1026>;expires=3600, <sip:a@172.25.25.83:5060>;expires=2868 Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 172.25.25.84:5060 "Noisy feedback tells: pid=21681 req_src_ip= 172.25.25.83 req_src_port=1025 in_uri=sip:172.25.25.84:5060 out_uri=sip: 172.25.25.84:5060 via_cnt==1" source=172.25.25.84:5060 to=172.25.25.83:1025 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.0.2:5060 ;rport=1025;branch=z9hG4bK600315716;received=172.25.25.83 From: <sip:a@192.168.0.2:5060>;tag=658770091 To: <sip:a@192.168.0.2:5060>;tag=b27e1a1d33761e85846fc98f5f3a7e58.b319 Call-ID: 2091069424 [!at] 192.168.0.2 (replace the [!at] with a @) CSeq: 1 REGISTER Contact: <sip:a@172.25.25.83:1026>;expires=3600, <sip:a@172.25.25.83:5060>;expires=2868 Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 172.25.25.84:5060 "Noisy feedback tells: pid=21682 req_src_ip= 172.25.25.83 req_src_port=1025 in_uri=sip:172.25.25.84:5060 out_uri=sip: 172.25.25.84:5060 via_cnt==1" source=172.25.25.88:5060 to=172.25.25.84:5060 INVITE sip:172.25.25.84:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport;branch=z9hG4bK1784363542 Route: <sip:a@192.168.0.2:5060> From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 5 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=172.25.25.88:5060 SIP/2.0 100 trying -- your call is important to us Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 172.25.25.84:5060 "Noisy feedback tells: pid=21677 req_src_ip= 172.25.25.88 req_src_port=5060 in_uri=sip:172.25.25.84:5060 out_uri= sip:a@192.168.0.2:5060 via_cnt==1" source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=192.168.0.2:5060 INVITE sip:a@192.168.0.2:5060 SIP/2.0 Record-Route: <sip:172.25.25.84;ftag=1669898773;lr=on> Via: SIP/2.0/UDP 172.25.25.84;branch=z9hG4bK21ea.9b53c1e3.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060> Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Contact: <sip:c@172.25.25.88:5060> Max-Forwards: 4 User-Agent: eXosip/0.1 Subject: hello Expires: 120 Supported: 100rel Allow: INVITE, ACK, UPDATE, INFO, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 264 Route: <sip:172.25.25.84:5060> P-hint: rr-enforced v=0 o=josua 0 0 IN IP4 172.25.25.88 s=conversation c=IN IP4 172.25.25.88 t=0 0 m=audio 5071 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 a=rtcp:5072 IN IP4 172.25.25.88 a=direction:passive source=172.25.25.84:5060 to=172.25.25.88:5060 cSIP/2.0 408 Request Timeout Via: SIP/2.0/UDP 172.25.25.88:5060;rport=5060;branch=z9hG4bK1784363542 From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cbaa Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 INVITE Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 172.25.25.84:5060 "Noisy feedback tells: pid=21691 req_src_ip= 172.25.25.88 req_src_port=5060 in_uri=sip:172.25.25.84:5060 out_uri= sip:a@192.168.0.2:5060 via_cnt==0" source=172.25.25.88:5060 to=172.25.25.84:5060 ACK sip:172.25.25.84:5060 SIP/2.0 Via: SIP/2.0/UDP 172.25.25.88:5060;rport;branch=z9hG4bK1784363542 Route: <sip:a@192.168.0.2:5060> From: <sip:c@172.25.25.88>;tag=1669898773 To: <sip:a@192.168.0.2:5060>;tag=a6a1c5f60faecf035a1ae5b6e96e979a-cbaa Call-ID: 1721341700 [!at] 172.25.25.88 (replace the [!at] with a @) CSeq: 20 ACK Content-Length: 0 --------------------------------------------------------------------------------------------------------------------------------------------- the SER configuration file is: ----------------------------------------------------------------- # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line #modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } --------------------------------------------------------------- Please help me in this regard. Asif

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[Serusers] problems installing SER
by Davidson, Michael D 22 Feb '06

22 Feb '06

I was running an older version of SER and decided to upgrade my installation. I downloaded a solaris 10 installation and did the pkgadd -d to install it. Everything seems to have worked, but when I try and start ser using serctl start I get the following response: Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed I can't seem to get past this point. Why does it care if the ser.pid file exists since it is supposed to be starting the application? I would appreciate any suggestions?

2 1

[Serusers] undefined symbol: db_free_query
by jvr_78＠yahoo.com.ar 22 Feb '06

22 Feb '06

Hi, I have this error, undefined symbol: db_free_query I change the version of ser, 0.8.14 to ser-0.10.99-dev30-tm-timers-pa-3 what is the funcion to replace db_free_query ? best regards

2 1

[Serusers] Re: [Sems] Sems Thread Memory
by Nicky 22 Feb '06

22 Feb '06

Hi Don, Thank you for your reply. How can I check using commands that the thread is still in session? What are the commands in the scripts that close the rtp session? Please advice. Thanks Regards, Nicky ----- Original Message ----- From: don.fletcher(a)att.net To: Nicky Sent: Tuesday, February 21, 2006 2:47 AM Subject: Re: [Sems] Sems Thread Memory Nicky, Probably the RTP streams are not being closed down properly at the end of a call so that the threads remain in existence. Look at the media streams with ethereal. Look closely at call disconnect. Be sure all threads are closed. don -------------- Original message from "Nicky" <nicky(a)caliber.com.sg>: -------------- Dear All, I am having a serious problem, my thread memory size is growing to up to 2GB which applies to SEMS, SER and Mysqld thread. Please help. I am testing with about 10 people calling in at the same time. What shall I do? :( Nicky ----- Original Message ----- From: Nicky To: serusers(a)lists.iptel.org ; sems(a)lists.iptel.org Sent: Saturday, October 29, 2005 11:44 AM Subject: pthread create failed with code 11 Hi Support, I have experienced this error, please advice what can be the caused. Thanks. ERROR: start (AmThread.cpp:85): pthread create failed with code 11 Regards, Nicky

1 0

RE: [Serusers] Authentication of ReInvite
by Vitaly Nikolaev 21 Feb '06

21 Feb '06

What about loose_route ? ReINVITE must be catched by sloose_route and... for example in my config almost in the beginning, before all kind of auth checks: if (loose_route()) { if(method == "INVITE") { xlog("L_DBG", "%ci: route, loose_route said yes for INVITE msg\n"); route(1); break; } else { xlog("L_DBG", "%ci: route, loose_route said yes for non INVITE msg\n"); t_relay(); break; } }; Then on route(1) there is auth.. (it was before in main route) I do usual nat stuff and forward the call to whenever Route: field pinted: t_on_reply("1"); if (! loose_route()) { if ((!lookup("location") && method == "INVITE" && uri == myself) || uri == myself) { sl_send_reply("404", "Not Found"); break; }; } if(method == "INVITE") { /* Handle re-INVITEs */ if (force_rtp_proxy("l")) { t_on_reply("2"); } else { if(force_rtp_proxy("")) t_on_reply("2"); }; }; if (!t_relay()) { sl_reply_error(); }; So, ReINVITE will be catched in forst loose_route if and bypass auth section will be sent to route(1) where it will not do lookup(location) but send wherever route field pointed. I am wrong ? :) -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Pat wang Sent: Tuesday, February 21, 2006 10:36 AM To: klaus.mailinglists(a)pernau.at Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] Authentication of ReInvite Hi Klaus, The /var/log/messages does not show any thing but here is the ngrep of the Re-Invite. I couldn't find anything wrong in the ACK for 407. Do you have any other thoughts? Thanks, Pat U 192.10.1.13:5060 -> 192.10.1.2:5060 INVITE sip:2101@192.10.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.10.1.13:5060;branch=z9hG4bK9523f072DDFFC805. From: "2103" <sip:2103@test.com:5060>;tag=9783CE70-D2E8B695. To: <sip:2101@test.com:5060>;tag=001280b9d20f80a2448c1c57-55a21cf1. Route: <sip:192.10.1.2;ftag=9783CE70-D2E8B695;lr=on>;ftag=9783CE70-D2E8B695;lr= on>. CSeq: 2 INVITE. Call-ID: 688c36b4-5bb67582-27524277(a)192.10.1.13. Contact: <sip:2103@192.10.1.13:5060>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0. Supported: 100rel,timer,replace. Allow-Events: talk,hold,conference. Max-Forwards: 70. Content-Type: application/sdp. Content-Length: 197. . v=0. o=- 1137681804 1137681804 IN IP4 192.10.1.13. s=Polycom IP Phone. c=IN IP4 192.10.1.13. t=0 0. m=audio 2252 RTP/AVP 0 101. a=sendonly. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. U 192.10.1.2:5060 -> 192.10.1.13:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.10.1.13:5060;branch=z9hG4bK9523f072DDFFC805. From: "2103" <sip:2103@test.com:5060>;tag=9783CE70-D2E8B695. To: <sip:2101@test.com:5060>;tag=001280b9d20f80a2448c1c57-55a21cf1. CSeq: 2 INVITE. Call-ID: 688c36b4-5bb67582-27524277(a)192.10.1.13. Proxy-Authenticate: Digest realm="test.com", nonce="43fb2e105733df37da780239bc94922da01a4177". Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 192.10.1.2:5060 "Noisy feedback tells: pid=26457 req_src_ip=192.10.1.13 req_src_port=5060 in_uri=sip:2101@192.10.1.11:5060 out_uri=sip:2101@192.10.1.11:5060 via_cnt==1". . U 192.10.1.13:5060 -> 192.10.1.2:5060 ACK sip:2101@192.10.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.10.1.13:5060;branch=z9hG4bK9523f072DDFFC805. From: "2103" <sip:2103@test.com:5060>;tag=9783CE70-D2E8B695. To: <sip:2101@test.com:5060>;tag=001280b9d20f80a2448c1c57-55a21cf1. Route: <sip:192.10.1.2;ftag=9783CE70-D2E8B695;lr=on>;ftag=9783CE70-D2E8B695;lr= on>. CSeq: 2 ACK. Call-ID: 688c36b4-5bb67582-27524277(a)192.10.1.13. Contact: <sip:2103@192.10.1.13:5060>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0. Max-Forwards: 70. Content-Length: 0. . U 192.10.1.2:5060 -> 192.10.1.11:5060 ACK sip:2101@192.10.1.11:5060 SIP/2.0. Record-Route: <sip:192.10.1.2;ftag=9783CE70-D2E8B695;lr=on>. Via: SIP/2.0/UDP 192.10.1.2;branch=0. Via: SIP/2.0/UDP 192.10.1.13:5060;branch=z9hG4bK9523f072DDFFC805. From: "2103" <sip:2103@test.com:5060>;tag=9783CE70-D2E8B695. To: <sip:2101@test.com:5060>;tag=001280b9d20f80a2448c1c57-55a21cf1. CSeq: 2 ACK. Call-ID: 688c36b4-5bb67582-27524277(a)192.10.1.13. Contact: <sip:2103@192.10.1.13:5060>. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER. User-Agent: PolycomSoundPointIP-SPIP_500-UA/1.3.0. Max-Forwards: 16. Content-Length: 0. P-hint: rr-enforced. . >From: Klaus Darilion <klaus.mailinglists(a)pernau.at> >To: Pat wang <wangyu39(a)hotmail.com> >CC: serusers(a)lists.iptel.org >Subject: Re: [Serusers] Authentication of ReInvite >Date: Tue, 21 Feb 2006 13:04:59 +0100 > >Pat wang wrote: > >>Morning everyone, >> >>Has anyone tried authentication of Re-INVITE on SER? When I add the >>proxy_challenge to the ReInvite in the SER configure file, the ACK for 407 >>from the client is porxied by SER to the other side. The signalling looks >>like: >> >>caller----------------------SER---------------------Callee >>------------------ normal call setup-------------------- >>--------------------------blha blha----------------------- >>------------------------------------------------------------- >>---call hold Re-Invite---> >><--- 407 challenge------ >>--------ACK-----------------> >> -------------ACK---------> *** This is >>not to be proxied! >>----ReInvite---------------> >>----------------------etc etc............ >> >>Anyone seen this while doing similar thing on SER? How can I stop SER >>proxing this ACK just like the way SER treat the original Invite with >>authentiacion? >> >>Here is the authentication part in the loose route block of my SER config >>file: >> >> if (method=="INVITE") { >> if (!proxy_authorize("test.com", "subscriber")) { >> proxy_challenge( "test.com", "0"); >> break; >> }; >> }; > >AFAIK, proxy challenge sends a stateless reply. ACK to stateless replies >should be absorbed by ser immediately without entering the ser.cfg routing >script. Thus, probably ser can't detect this ACK as stateless. > >Please watch syslog (debug=4 and "tail -f /var/log/syslog|grep -v qm_") >and verify why ser does nto detect a "stateless" ACK. > >Also verifiy the ACK if all the tags are correctly copied from 40x to the >ACK. Maybe this is cause by the branch=0 bug? > >regards >klaus _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Users] new textops functions
by Daniel-Constantin Mierla 21 Feb '06

21 Feb '06

Hello, the development version of textops module has a new set of functions that work on the body of the SIP messages. - search_body() - search a regular expression within the message's body - search_append_body() - search a regular expression within the message's body and append a text after it - replace_body() - replace a regular expression match with a text - replace_body_all() - replace all regular expression matches with a text - subst_body() - sed/perl like subst operation over the message's body More details about these functions at: http://openser.org/docs/modules/1.1.x/textops.html Cheers, Daniel

2 1

[Serusers] Authentication of ReInvite
by Pat wang 21 Feb '06

21 Feb '06

Morning everyone, Has anyone tried authentication of Re-INVITE on SER? When I add the proxy_challenge to the ReInvite in the SER configure file, the ACK for 407 from the client is porxied by SER to the other side. The signalling looks like: caller----------------------SER---------------------Callee ------------------ normal call setup-------------------- --------------------------blha blha----------------------- ------------------------------------------------------------- ---call hold Re-Invite---> <--- 407 challenge------ --------ACK-----------------> -------------ACK---------> *** This is not to be proxied! ----ReInvite---------------> ----------------------etc etc............ Anyone seen this while doing similar thing on SER? How can I stop SER proxing this ACK just like the way SER treat the original Invite with authentiacion? Here is the authentication part in the loose route block of my SER config file: if (method=="INVITE") { if (!proxy_authorize("test.com", "subscriber")) { proxy_challenge( "test.com", "0"); break; }; }; Thanks, Pat

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[Serusers] avpops and wav file
by amellado＠uct.cl 21 Feb '06

21 Feb '06

Hi' I want to use voicemail in serweb. I've avpops working, but how I send the voicemail wav file to voice_silo table. Thanks.

1 0

[Serusers] Users in ENUM
by Anderson Alves Albuquerque 21 Feb '06

21 Feb '06

If I have a prefix 1100 and I send call to 11000202, how can I use ENUM this? If I have the line below, I would need to put username but I don´t want this: $ORIGIN 0.2.0.0.0.1.1.e164.arpa. 2 IN NAPTR 1 10 "u" "E2u+sip" "!^.*$!sip:10.10.10.1<r(a)10.10.10.1>; sip:john@10.10.10.2!" . Suppose, I send to 11000202 or 11000203 I want send call to 10.10.10.2 but I want to use the e164 original. If have a lot of users I would need to put all the users in ENUM, but I don´t want this. How can I do this?

1 0

[Serusers] Forwarding to port 5060 when UA behind NAT
by Arek Bekiersz 21 Feb '06

21 Feb '06

Dear list, I have situation where SIP UA is behind NAT, that maps everything to ports above number 61000. I use nathelper, rtpproxy and SER 0.8.12. Everything works fine. Until I have UA that does not send "Contact" field in BYE (or OK for BYE) during call termination. In this case SER is forwarding responses not to NAT-ed port, but to port 5060. QUESTION: What can be done to correct this situation? Is this improved in new SER releases? Sequence is as follows: IP_of_A:61001 -> IP_of_proxy:5060 BYE sip:B@IP_of_proxy loose_route ... IP_of_proxy:5060 -> IP_of_B:5060 BYE sip:B@IP_of_B ... IP_of_B:5060 -> IP_of_proxy:5060 SIP/2.0 200 OK ... IP_of_proxy:5060 -> IP_of_A:5060 <WRONG! Should be 61001> SIP/2.0 200 OK ... It does not happen when terminating UA sends "Contact" field in first BYE. -- Regards, Arek Bekiersz

1 1

[Users] TLS stability
by Klaus Darilion 21 Feb '06

21 Feb '06

Hi! Recently on the serdev mailinglists there were considerations about the stability of the TLS code: http://mail.iptel.org/pipermail/serdev/2006-January/006608.html I know there was a fix to disable openssl compression. Was this the only known problem? Is TLS now running stable? Has someone ever used the TLS stack successfully under load? regards klaus

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Re: [Serusers] RV: serweb error
by Karel Kozlik 21 Feb '06

21 Feb '06

HI, set fifo_mode = 666 in your ser.cfg Karel Guty napsal(a): > > > Hello! > > > > I’m having a trouble with my serweb. > > > > I can access to the user interface login screen but when I log on I > receive the next error: > > > > *Warning*: fopen(/tmp/ser_fifo): failed to open stream: Permission > denied in */home/gustavo/ser/serweb-0.9.4/html/functions.php* on line *340* > > *Warning*: Cannot modify header information - headers already sent by > (output started at > /home/gustavo/ser/serweb-0.9.4/html/functions.php:340) in > */home/gustavo/ser/serweb-0.9.4/html/page.php* on line *35* > > * * > > Thanks!!!!!! > > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] RV: serweb error
by Guty 21 Feb '06

21 Feb '06

Hello! I'm having a trouble with my serweb. I can access to the user interface login screen but when I log on I receive the next error: Warning: fopen(/tmp/ser_fifo): failed to open stream: Permission denied in /home/gustavo/ser/serweb-0.9.4/html/functions.php on line 340 Warning: Cannot modify header information - headers already sent by (output started at /home/gustavo/ser/serweb-0.9.4/html/functions.php:340) in /home/gustavo/ser/serweb-0.9.4/html/page.php on line 35 Thanks!!!!!!

1 0

[Users] caller number removal
by unplug 21 Feb '06

21 Feb '06

In the situation below, a call made from PSTN to openser. PSTN -----> openser ---------> UA1 How can I remove the caller such that UA1 can't show the caller number in her display?

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[Users] preparation of patch release 1.0.1
by Daniel-Constantin Mierla 21 Feb '06

21 Feb '06

Hello everybody, a new package building of CVS branch rel_1_0_0 (release v1.0.0) will take place soon. This comes as an update to release 1.0.0 which will include the fixings occurred since 1.0.0 was released. We like to collect all issues of v1.0.0 you are aware of, to be fixed before package generation. Next is a list of the issues that should be fixed: - TLS - when TLS connections is refused, immediate reply instead of waiting for timeout - TCP - when the call is initiated over TCP but the contact advertised is UDP, the BYE from callee is not routed properly (it is tried TCP due to double RR) - record_route_preset() parameter size - ruri matching of canceled transactions (see: http://openser.org/pipermail/users/2006-February/002951.html) Any feedback is appreciated. Cheers, Daniel

5 8

RV: [Users] Multiple Interfaces rtpproxy
by Fernando Rodriguez 21 Feb '06

21 Feb '06

Could a solution be using multiple domains per interface of the machina so i can run diferent instantes of the media proxy ?? _____ De: users-bounces(a)openser.org [mailto:users-bounces@openser.org] En nombre de Fernando Rodriguez Enviado el: Lunes, 20 de Febrero de 2006 11:32 a.m. Para: users(a)openser.org Asunto: [Users] Multiple Interfaces rtpproxy Hello, I have a running openser on a machine with multiple interfaces im using mediaproxy but I was only able to set it up using 2 interfaces .. How can I set it up to have more working interfaces 1 lan 1 dmz 1 wan Right now it only mediaproxies the wan and dmz interfaces But I want to be able to mediaproxy betrwen the lan and wan and dmz to lan Does any one has a set up like this and will be able to guide me on seting this up .. Thanks a lot for your help Fernando Rodriguez V. frod(a)aitelecom.net

2 1

[Users] Parameter domain_prefix in AVPOPS
by Papadopoulos Georgios 21 Feb '06

21 Feb '06

Hello, Just as a sugestion, wouldn't it be useful to add a parameter like "domain_prefix" in AVPOPS? thank you George

2 1

[Users] add/rewrite a tag/parameter in SDP
by Lenir 21 Feb '06

21 Feb '06

Hey guys, What is the best way to modify the SDP message to always include (if not replace) the a= tags in a SDP message? My problem is specifically with g729, according to the RFC, if the parameter: "a=fmtp:18 annexb=yes" is present or missing then it is assumed that g729b (g729 with VAD) and if "a=fmtp:18 annexb=no" is present then the devices will use g729a (g729 with VAD Disabled). My problem is that some cisco gateways and ATAs behave differently (of course Cisco always doing it their way), if "a=fmtp:18 annexb=no" is missing then it assumes that g729a will be used, instead of following the RFC where it should negotiate g729b. Also, asterisk doesn't support g729b so I want to avoid using g729b everywhere in my network. So what is the best way or is there a way, to either: a) insert "a=fmtp:18 annexb=no" if is not present b) replace "a=fmtp:18 annexb=yes" for "a=fmtp:18 annexb=no" if it is present Thanks in advance!! Lenir p.s. Please see the below excerpt from RFC 3555 regarding how to specify G729. RFC 3555 MIME Type Registration of RTP Payload Formats July 2003 4.1.9. Registration of MIME media type audio/G729 MIME media type name: audio MIME subtype name: G729 Required parameters: None Optional parameters: ptime, maxptime annexb: indicates that Annex B, voice activity detection, is used or preferred. Permissible values are "yes" and "no" (without the quotes); "yes" is implied if this parameter is omitted

4 4

[Users] non default port setting
by unplug 21 Feb '06

21 Feb '06

I have 2 questions about the port using in openser with NAT enabled. Q1. Can I using another port number instead of 5060? I have tried to change the setting in configuration file. Below is a part of configuration file with NAT enabled. ... port=6060 ... if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("203.193.26.234:6060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; Sip phone (UA1) can register in openser successfully. But in the location table, I have the contact and received field still using port 5060. sip:871966806561@10.0.0.46:5060 | sip:210.184.23.31:5060 As a result, UA1 failed to make a call as the port is not the one we want. Q2. How can I set multiple port for an openser if another port can be used instead of 5060? Say I want openser listen to 3 ports, ie 5060, 6060, 7060.

2 1

[Users] timeout problem with media gateway
by Matteo Piazza 21 Feb '06

21 Feb '06

HI, In my serial forking configuration I have this problem: if i use the a media gateway ( asterisk ) on the same machine ( but I see also on the another machine) of openser ( on the 5061 port) I have a problem with the timeout of a call. The call continues beyond the timeout and when by the PSTN telephone I reject the call appears the following log message: Warning: sl_send_reply: I won't send a reply for ACK!! Where I have mistake? Any idea? Another thing. I see that I redirect the call directly in the media gateway, (for example a location of a user is a PSTN number) whitout the traslation of IP number, all is ok. Matteo ====================================== # # $Id: openser.cfg$ # # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/lcr.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("registrar", "append_branches", 1) #modparam("usrloc", "db_mode", 0) # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("usrloc","db_url","mysql://openser:heslo@127.0.0.1/openser") modparam("auth_db","db_url","mysql://openser:heslo@127.0.0.1/openser") modparam("lcr","db_url","mysql://openser:heslo@127.0.0.1/openser") modparam("avpops","avp_url","mysql://openser:heslo@127.0.0.1/openser") modparam("avpops","avp_table","usr_preferences") modparam("tm", "fr_timer", 7) modparam("tm", "fr_inv_timer", 10) modparam("tm", "wt_timer", 5) #modparam("avpops","avp_aliases","fwdbusy=i:665") # ------------------------- request routing logic ------------------- # main routing logic route{ # ============================================== # initial sanity checks -- messages with # max_forwards==0, or excessively long requests # ============================================== if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # ============================= # Notify Keep-Alive Section # ============================ if ((method=="NOTIFY") && search("Event: keep-alive")) { sl_send_reply("200","OK"); exit; }; if (method =="INVITE" && uri =~"^sip:0[0-9]*@*"){ log(1, "Check 1 Start PSTN Call\n"); rewritehostport("192.168.9.97:5061"); } # ======================================================== # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol #======================================================== if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if(method=="BYE"){ t_relay(); exit; }; # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay_to_tls("IP_domain1","port_domain1"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay_to_tls("IP_domain2","port_domain2"); # exit; #} route(1); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("create-net.it", "subscriber")) { www_challenge("create-net.it", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; # ======================================= # PSTN Getaway # ======================================= if(method=="INVITE"){ if (avp_db_load("$ruri","s:fwdactive")) { if(avp_check("s:fwdactive","eq/y/i")){ log(1,"FWD ACTIVE\n!"); setflag(2); } }; if (avp_db_load("$ruri","s:mailactive")) { if(avp_check("s:mailactive","eq/y/i")){ log(1,"MAIL ACTIVE\n!"); setflag(4); } } if(load_contacts()){ xlog("L_ERR","LOAD CONTACTS!\n"); setflag(1); }; if(next_contacts()){ xlog("L_ERR","NEXT CONTACT!\n"); }; }; if(isflagset(2)){ t_on_failure("1"); t_relay(); exit; } else { if(isflagset(4)){ t_on_failure("2"); t_relay(); exit; } else { t_on_failure("3"); t_relay(); exit; } }; } route[1] { # log(1,"ROUTE 1 !\n"); if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP log(1,"ROUTE 2 !\n"); sl_send_reply("181","CALL IS BEING FORWARDED"); t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1]{ log(1,"FAILURE ROUTE FORWARD !\n"); if (t_check_status("408")){ if (next_contacts()) { log(1,"NUOVO CONTATTO !\n"); route(2); exit; } else { log(1,"FINE CHIAMATA - TIME OUT!\n"); if (isflagset(4)){ log(1,"ATTIVAZIONE VOICEMAIL!\n"); revert_uri(); xlog("L_ERR","<$ruri>"); rewritehostport("192.168.9.97:5061"); append_branch(); t_relay(); exit; } exit; }; } else { if ( t_check_status("603") || t_check_status("486") ){ log(1,"OCCUPATO!\n"); if (isflagset(4)){ log(1,"ATTIVAZIONE VOICEMAIL!\n"); revert_uri(); xlog("L_ERR","<$ruri>"); rewritehostport("192.168.9.97:5061"); append_branch(); t_relay(); exit; } exit; } }; } failure_route[2] { log(1,"FINE CHIAMATA 2 - SENZA FORWARD!\n"); revert_uri(); xlog("L_ERR","<$ruri>"); rewritehostport("192.168.9.97:5061"); append_branch(); t_relay(); exit; } failure_route[3] { log(1,"FINE CHIAMATA 3 - SENZA FORWARD!\n"); if ( t_check_status("603") || t_check_status("486") ){ t_relay(); exit; } exit; } ======================================= Matteo Piazza, Junior Researcher CREATE-NET Via Solteri, 38 - 38100 Trento - Italy email: matteo.piazza(a)create-net.it Tel: +39-0461-408400ext:308 www.create-net.it =======================================

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[Serusers] avp_radius
by samuel 21 Feb '06

21 Feb '06

Hi all!! I'm testing stable ser version (0.9.6) with RADIUS authentication. I have properly setup the auth and next step was to get some extra attributes from the auth server using the avp_radius module. The thing is that I am a newby in the RADIUS world and don't know exactly what do I have to setup in both dictionaries. The main problem is that the attribute I have to read from ser script is a cisco-h323-whatever and don't even know if it is possible to read it as a caller_cisco-h323-whatever ...I wish I could use another attribute but I can not modify it :( I have set in both dictionaries cisco vendor attribute, the cisco-h323 attribute, SIP_AVP, SIP-caller-AVPs, and SIP_Callee-AVPs. I guess there something missing to glue the AVP with the cisco staff (if possible). ser log just says: avp_load_user: Failure DEBUG:avpops:check_avp: no avp found to check Any feedback is highly appreciated (even links to documentation,howtos....) Thank you, Sam.

2 2

Re: [Users] Beer at wednesday 22 at 18:00 ?
by Helge Waastad 21 Feb '06

21 Feb '06

Hi, yet again a brilliant initiative, Atle. As much as I would like to join this time, I just cant. I hope we can have another go again soon. br hw -- Helge Waastad Senior Konsulent Systemavdelingen Smartnet

1 0

[Users] how to use OpenSER
by Brijesh 21 Feb '06

21 Feb '06

hi all i am explaining my problem i have installed openser on one of machine. i have assigned SIP_DOMAIN in that machine as "test.com". then i also added users to the openser using *openserctl add user1(a)test.com test user1(a)test.com* command. so that user is also added to the myslql database. till now it is ok. now i am using Minisip as a sip end application. in that i have given account name as user1(a)test.com, sip uri as sip:user1@test.com, sip proxy address as machine's ip address on which proxy is installed and network port as 5060. but it is not working.. can someone help me out.. thanx in advance. Brijesh

2 1

[Serusers] Help! DNS?
by ryan nalupa 21 Feb '06

21 Feb '06

hello! i was just wondering, since i'm still new here, where will i config this? i'm using debian sarge amd64 and ser 0.9.4. DNS SVR Resource Records It is important that your SIP clients can connect to your server for purposes of registration and call control. You might even want to have a redundant server to handle calls if your primary server is unavailable. These requirements can be meet by using DNS SVR Resource Records, available in BIND 8.X and up releases. The format for a SVR RR is this: _service._protocol SVR Priority Weight Port hostname In this case we want to establish an entry for our primary SIP server, gateway.mydomain.com, that will listen on UDP port 5060. The entry will look like this: _sip._udp SRV 0 0 5060 gateway.mydomain.com Placement of the new resource record is important. Here is a sample zone file: ; zone 'mydomain.com' last serial 1998071308 $ORIGIN com. mydomain 86400 IN SOA gateway.mydomain.com. postmaster.mydomain.com. ( 1998111908 ; Serial 36000 ; Refresh 900 ; Retry 36000 ; Expire 28800 ); Minimum IN NS gateway.mydomain.com. IN NS ns3.backupdomain.com. IN MX 1 gateway.mydomain.com. IN A 192.168.0.1 ;If we place the SRV record above the next line it fails to load $ORIGIN fitawi.com. _sip._udp SRV 0 0 5060 gateway.mydomain.com. gateway IN A 192.168.0.1 www IN CNAME gateway.mydomain.com. After reloading your zone file you can verify that the entry is working by using dig. dig -t SRV _sip._udp.mydomain.com The results should look something like this: ; <<>> DiG 9.1.0 <<>> -t SRV _sip._udp.mydomain.com ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 32654 ;; flags: qr aa rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 2, ADDITIONAL: 1 ;; QUESTION SECTION: ;_sip._udp.mydomain.com. IN SRV ;; ANSWER SECTION: _sip._udp.mydomain.com. 86400 IN SRV 0 0 5060 gateway.mydomain.com. ;; AUTHORITY SECTION: mydomain.com. 86400 IN NS ns3.elsewhere.com. mydomain..com 86400 IN NS gateway. mydomain.com. ;; ADDITIONAL SECTION: gateway. mydomain.com. 86400 IN A 192.168.0.150 ;; Query time: 6 msec ;; SERVER: 192.168.0.150#53(192.168.0.150) ;; WHEN: Tue Dec 3 08:34:17 2002 ;; MSG SIZE rcvd: 132 thanks in advance! ryan --------------------------------- Do you Yahoo!? Try the new Yahoo! Philippines Front Page!

2 1

[Serusers] loose_route bug?!
by Ramin Nikaeen 21 Feb '06

21 Feb '06

Hi Everyone, I am using SER version Server: Sip EXpress router (0.9.2 (i386/linux)). I am initiating a call from a caller sip UA to a callee sip UA and I am receiving the ACK response to the INVITE from the callee on SER. Before loose_route processing it looks like this: ACK sip:2224440133@10.1.10.65:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.15.52:5060;rport;branch=z9hG4bK83e54f590401584628d473053d1ef8a8 From: <sip:2224440132@xxx.yyy.zzz:5060>;tag=1cac2190 To: <sip:2224440131@xxx.yyy.zzz:5060>;tag=d323038ba423a329i0 Call-ID: 66cb6ba85191e6025ebde2c552177442(a)172.16.15.52 CSeq: 1 ACK Contact: <sip:2224440132@172.16.15.52:5060> Max-Forwards: 70 Route: <sip:10.1.10.65;ftag=1cac2190;lr>,<sip:10.1.10.65;ftag=1cac2190;lr> Content-Length: 0 After loose_route processing and relaying, the ACK, instead of being routed to the callee (2224440133) User Agent, it is received back at SER looking like this: ACK sip:10.1.10.65;ftag=1cac2190;lr SIP/2.0 Record-Route: <sip:10.1.10.65;ftag=1cac2190;lr=on> Via: SIP/2.0/UDP 10.1.10.65;branch=0 Via: SIP/2.0/UDP 172.16.15.52:5060;rport=5060;branch=z9hG4bK83e54f590401584628d473053d1ef 8a8 From: <sip:2224440132@xxx.yyy.zzz:5060>;tag=1cac2190 To: <sip:2224440131@xxx.yyy.zzz:5060>;tag=d323038ba423a329i0 Call-ID: 66cb6ba85191e6025ebde2c552177442(a)172.16.15.52 CSeq: 1 ACK Contact: <sip:2224440132@172.16.15.52:5060> Max-Forwards: 16 Route: Content-Length: 0 Why is SER routing this ACK request incorrectly?! Is the syntax of the original ACK generated by the Callee's UA incorrect?! I would appreciate any help as I can't figure it whether there is something wrong with ACK's syntax or there is a bug in SER's loose_route processing?! Thanks a lot ramin

2 1

RE: [Users] next_gw(): No ruri_user AVP
by Raymond Chen 21 Feb '06

21 Feb '06

Hi Bogdan, I solved problem by forcing avp_write("$ruri/username", "i:1402"). Thanks Ray -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Raymond Chen Sent: Thursday, February 16, 2006 12:27 AM To: 'Bogdan-Andrei Iancu' Cc: Users(a)openser.org Subject: RE: [Users] next_gw(): No ruri_user AVP Hi Bogdan, Both, after load_gw and before next_gw avps are visible. But not after next_gw. There is no where in the script delete any avp. Ray -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Wednesday, February 15, 2006 8:18 AM To: Raymond Chen Cc: daniel(a)voice-system.ro; Users(a)openser.org Subject: Re: [Users] next_gw(): No ruri_user AVP Ray, I think there is a problem in the script - if you see the gw_uri avps, it's quite impossible not to see them also before and after the next_gw() call in request route. Disregarding the ruri_avp, the gw_uri avps should be visible. are you sure you do not delete all the avps in your script. Try placing avp_print before and after each lcr function call. regards, bogdan Raymond Chen wrote: >Hi bogdan, > >I put avp_print after next_gw , no avp output. > > >1(1334) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.138:5060> > 1(1334) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.139:5060> > 1(1334) does_uri_exit(): User in request uri does not exist > 1(1334) is_user_in(): User is in group 'local' > 1(1334) db_flags=3, flags=12 > 1(1334) DEBUG:avpops:load_avps: loaded avps = 1 > 1(1334) parse_headers: flags=ffffffffffffffff > 1(1334) DEBUG:avpops:pushto_avps: 1 avps were processed >1(1334) next_gw(): No ruri_user AVP > > > > > > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Wednesday, February 15, 2006 4:39 AM >To: Raymond Chen >Cc: daniel(a)voice-system.ro; Users(a)openser.org >Subject: Re: [Users] next_gw(): No ruri_user AVP > >Ray, > >the ruri_avp is added by next_gw after its usage from the REQUEST route. >You may check this by placing an avp_print after you did next_gw() in >request route (after calling route 3, for example). Check if there is >any avp with ID 1402. > >regards, >bogdan > >Raymond Chen wrote: > > > >>Hi bogdan, >> >>We called Load_gw and next-gw from request route. We have no problem with >>next_gw if the if (avp_pushto("$ruri", "s:fwdnoanswer")) happens in the >> >> >main > > >>route. But when it does in the failure_route the next_gw can't find the >>ruri_user avp. >> >>Raymond >> >> >> >>Route { >> >> .......... >> >> Route(3); >> >> ........... >> >>} >> >>failure_route[1] { >> >> if (t_check_status("(480)|(408)")) { >> if (avp_pushto("$ruri", "s:fwdnoanswer")) { >> avp_delete("s:fwdnoanswer"); >> route(3); >> }; >> }; >> >>} >> >>Route[3] { >> >> if (!load_gws()) { >> sl_send_reply("500", "Server Internal Error - Cannot load >>gateways"); >> return; >> }; >> >> ............... >> >> Route(5); >> >>} >> >>Route[5] { >> >> if (!next_gw()) { >> rewriteuri("sip:userbusy@211.102.91.134:443"); >> t_relay(); >> return; >> }; >> >> .............. >> >>} >> >> >> >>-----Original Message----- >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>Sent: Wednesday, February 15, 2006 2:25 AM >>To: Raymond Chen >>Cc: daniel(a)voice-system.ro; Users(a)openser.org >>Subject: Re: [Users] next_gw(): No ruri_user AVP >> >>Hi Ray, >> >>do you call load_gws() from failure route? if so, not that this is not >>supported. >>For LCR to work properly, you need to call load_gws() and next_gw() from >>the request route and later next_gw() from failure routes. >> >>regards, >>bogdan >> >>Raymond Chen wrote: >> >> >> >> >> >>>Hi Bogdan, >>> >>>Here is the debug >>> >>>1(2584) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.138:5060> >>>1(2584) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.139:5060> >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f167c8, flags=2 >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.139:5060> >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f16790, flags=2 >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.138:5060> >>>1(2584) does_uri_exit(): User in request uri does not exist >>>1(2584) is_user_in(): User is in group 'local' >>>1(2584) db_flags=3, flags=12 >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f167f8, flags=B >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.139:5060> >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f16790, flags=2 >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.xxx:5060> >>>1(2584) next_gw(): No ruri_user AVP >>> >>> >>> >>>Raymond >>> >>> >>>-----Original Message----- >>>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>>Sent: Tuesday, February 14, 2006 10:55 AM >>>To: Raymond Chen >>>Cc: daniel(a)voice-system.ro; Users(a)openser.org >>>Subject: Re: [Users] next_gw(): No ruri_user AVP >>> >>>Hi Ray, >>> >>>use avp_print() after the load_gw() to see what avps were loaded and >>>again just before next_gw() to see the available avp. >>>this will help to see if it's a problem at the load or search part. >>> >>>regards, >>>bogdan >>> >>> _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users

1 0

[Serusers] Errors...
by ryan nalupa 21 Feb '06

21 Feb '06

hello! i'm following the ser howto that i got at iptel.org and i'm having errors when following these... 3.2. Adding an admin for your realm Now that we have a working database and ser is configured to use it, we need to add some users and at least one of them should have administrator privileges. The administrator role becomes important if you want to use a web management tool such as serweb. Basic account manipulation can be performed with the serctl script, located in /usr/sbin. To add a user use these commands serctl add JoeUser qwerty joe(a)mydomain.com The system notify for "Type MySQL Password", the default password is "heslo" To make JoeUser an administrator, we need to logon to MySQL and modify the database. mysql> connect ser; mysql> update subscriber set perms=?admin? where USER_ID=?JoeUser?; Query OK, 1 row affected (0.00 sec) Rows matched: 1 Changed: 1 Warnings: 0 mysql> select * from subscriber; | 4cefa7a4d3c8c2dbf6328520bd873a19 | JoeUser | qwerty | | | | joe(a)mydomain.com | 2002-12-02 19:20:41 | 2002-12-02 20:29:46 | 80e0f273b2067d40277b49ff842bb9e3 | o | | | c79a8f8f08596baa84bb02c88884426d | mydomain.com | f322c94b8b2fbe557d43ab3ac9e05b3a | admin | 1 | America/Los_Angeles | The third from last field shows that Joe has been assigned admin privileges. At this point Joe can logon to our server, but since he is the only user, there is not much he can do. We can now add additional users using the serctl script, or now is a good time to look at installing serweb, which will allow users to subscribe to our service. sql gives me an error saying check the manual that came with my mysql dist. and says an error from ?admin? where USER_ID=?JoeUser?. can anyone help me here..... :( ryan --------------------------------- Do you Yahoo!? Try the new Yahoo! Philippines Front Page!

1 0

Re: [Serusers] Problem with time of call invite in Mysql
by mkumar＠mantragroup.com 21 Feb '06

21 Feb '06

Hi, Thanks for the reply. I can chage my TimeZone but many Users will be having many timezones and if calls of each and every user records in Mysql DB according to his timezone then how can we have a common timezone for all atleast for getting time records in a particular period for billing. Manoj. > Quoting Andrey Kouprianov <andrey.kouprianov(a)gmail.com>: > >> Change your timezone >> >> On 2/20/06, mkumar(a)mantragroup.com <mkumar(a)mantragroup.com> wrote: >>> Hi All, >>> >>> I have SER and mysql DB in USA and I am calling from India. For all >>> the calls I >>> am dialling from India the time of call invite is getting recorded >>> 5.5 hours >>> ahead the local time in USA. Suppose if time in USA is 6:30 and if >>> I call from >>> India, in mysql DB it is getting stored that I called at 12:00. So >>> for billing >>> if I calculate calls between NOW() and NOW()-30 days, all the calls >>> in this 5.5 >>> hours are missing. >>> Please tell me where I have to change my configuration. I have ser >>> -0.9.4 and >>> Mysql 4.0 on Linux. >>> >>> Thanks for your help and time, >>> Manoj. >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >> > > >

1 0

[Serusers] modifying SDP of 180
by Ever Zalazar 21 Feb '06

21 Feb '06

Hi, I have a question. I need to modify the SDP of code 180 when my clients use my mediaproxy. How is the process between a SER and the media proxy? Best REgards Ever

1 0

[Serusers] SER billing softwares
by mkumar＠mantragroup.com 20 Feb '06

20 Feb '06

Hi All, Are there any open source SER billing softwares? If there are any, please point me to that. Thanks, Manoj.

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[Users] Beer at wednesday 22 at 18:00 ?
by Atle Samuelsen 20 Feb '06

20 Feb '06

Hi guys, girls and Geeks :-) Have you booked your Plain-ticket's yet? No? Remember, it's beer meeting on Wednesday the 22 february 2006 at 18:00. The beer will be consumed at Beer Palace at Akerbrygge Oslo, Norway. If somebody wanna join in on IP-Conferance, We can try to arrange that. (pleace send a mail saying if you are coming, since if we are more then a few ppl we need to book a table) Best Regards Greger Teigre and Atle Samuelsen

4 5

[Serusers] SER load balancer for asterisk
by Script Head 20 Feb '06

20 Feb '06

Hello, I am trying to build a setup that's probably a very generic. I have a number of asterisk servers that I use to handle transcoding. What I'd like to use SER for is to "spread" calls among these asterisk servers. Where should I start looking? ScriptHead

3 2

[Serusers] permissions module & allow_routing
by Kostas Marneris 20 Feb '06

20 Feb '06

Hello all, I have some trouble with permissions module and allow_routing(). I want to allow_routing ONLY from IP 1.2.3.4 (which is my PSTN GW) and all the rest will be denied (actually all the other will be send for authentication with "radius_proxy_authorize"). What is the syntax of the permissions.* files ? The following doesn't work.. permissions.deny: ALL : ALL permissions.allow: "^sip:*@1.2.3.4" : "^sip:*@10.10.10.1" where 10.10.10.1 is my SER's IP. thanks for any help, Kostas -- Kostas Marneris e-mail: K.Marneris(a)otenet.gr

3 2

[Users] Mysql DB Auth
by Sam Lee 20 Feb '06

20 Feb '06

Hi all, I've just finish installing the OPENSER 1.0 Trying to make it work on the mysql part. Just can't get it to connect to my mysql server properly with the correct authentication. It seems like it tries to go in as the default userid/password (openserro,openser) even though i've specified an alternate userid password. modparam("auth_db|uri_db|usrloc|alias_db", "db_url", "mysql://ser:heslo@localhost/ser") Refer to above , this line is already included in the openser.cfg. As u might have see, i am trying to migrate to openser without having to create the database with all the users again. Please advise how to i troubleshoot. Regards, Sam

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[Users] Multiple Interfaces rtpproxy
by Fernando Rodriguez 20 Feb '06

20 Feb '06

Hello, I have a running openser on a machine with multiple interfaces im using mediaproxy but I was only able to set it up using 2 interfaces .. How can I set it up to have more working interfaces 1 lan 1 dmz 1 wan Right now it only mediaproxies the wan and dmz interfaces But I want to be able to mediaproxy betrwen the lan and wan and dmz to lan Does any one has a set up like this and will be able to guide me on seting this up .. Thanks a lot for your help Fernando Rodriguez V. frod(a)aitelecom.net

1 0

[Users] new version of "OpenSER Database Structure" document
by Daniel-Constantin Mierla 20 Feb '06

20 Feb '06

Hello, a new version of "OpenSER Database Structure" document submitted by Norman Brandinger is posted at: http://openser.org/docs/OpenSER_Database_Structure.html It describe the meaning of the columns in the tables used by openser. Once again, many thanks to Norman! Cheers, Daniel

1 0

[Serusers] Is it possible to run SerWeb and SER on separated machines? (what about FIFO?)
by benaudi＠gmx.de 20 Feb '06

20 Feb '06

Hello, is it at the moment possible to run SER and SerWeb on two different pc's? In SerWeg config-file you have to define the path to "ser_fifo". But i cant access the ser_fifo which is running on another PC. Is there any example or discussion already? Thank you in Advance! Alex -- Lust, ein paar Euro nebenbei zu verdienen? Ohne Kosten, ohne Risiko! Satte Provisionen f�r GMX Partner: http://www.gmx.net/de/go/partner

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[Serusers] error querying table version
by Yi Zheng 20 Feb '06

20 Feb '06

Hi, While trying to setup auth_db() with mysql, I got the following error messages in /var/log/messages. table_version(): Invalid number of rows received: 2, location Feb 16 17:55:17 localhost ./ser[13709]: register_udomain(): Error while querying table version Feb 16 17:55:17 localhost ./ser[13709]: domain_fixup(): Error while registering domain I used ser_mysql.sh create to generate the ser database. It looks like that ser has two location entries in its version table. I tried to use ser_mysql.sh to reinstall the ser database but it ended up the same. What should I do to correct this? - ming

2 1

[Serusers] Multiple domains via a DB
by Nick Hoffman 20 Feb '06

20 Feb '06

How do you setup a multi-domain SER using a database? I've seen examples where you hard code into ser.cfg a regular expression for each domain. Is there a call that can do something similar as in compare the URI with returned values from the database? I tried this: if(is_from_local()) { sl_send_reply("600", "is local"); break; }; but it didn't fire for existing or non-existing domains. Does anyone have suggestions as to what should be done instead, or what needs to be fixed? I've attached my ser.cfg in case it's of any help. Thanks for your help, -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

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[Serusers] UA Header Field & Mediaproxy
by George Lambson 20 Feb '06

20 Feb '06

Does anyone know how to insert a User-Agent field into the header BEFORE Mediaproxy gets it? I am trying to set up mediaproxy to work with a UA that is asymmetric, but it doesn't recognize the UA because this header field is missing. Thanks, George

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[Serusers] Postgres + ser + problem
by Andrew Nowrot 20 Feb '06

20 Feb '06

Hi I am new to ser issues. I installed it last Friday. I managed to set up it correctly with Asterisk, but now I have a slightly problem with authorization, frankly with postgres. part of my ser.cfg file looks like this: # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/postgres.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) modparam("usrloc", "user_column", "username") modparam("usrloc","db_url", "postgres://ser:ser1234@localhost:5432/ser") # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # -- auth params -- # Uncomment if you are using auth module # modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # modparam("auth_db", "password_column", "password") modparam("auth_db","db_url", "postgres://ser:ser1234@localhost:5432/ser") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("ser.leon.pl", "subscriber")) { www_challenge("ser.leon.pl", "0"); break; }; save("location"); break; }; } *** my postgres table subscriber look like this: username | domain | password ------------ + ------------- + ---------- 2222 | ser.xxx.org | serxxx this is my only postgres table connected with ser. I did my configuration according to http://www.voip-info.org/wiki-SER+module+postgres. When I restarting ser I getting this error in syslog and my phones are not able to register to ser: Feb 18 16:31:39 asterix /usr/sbin/ser[6164]: Maxfwd module- initializing Feb 18 16:31:39 asterix /usr/sbin/ser[6164]: PG[364] submit_query query 'select table_version from version where table_name='subscriber'', result 'ERROR: relation "version" does not exist ' Feb 18 16:31:39 asterix /usr/sbin/ser[6164]: db_query(): Error while submitting query Feb 18 16:31:39 asterix /usr/sbin/ser[6164]: table_version(): Error in db_query Feb 18 16:31:39 asterix /usr/sbin/ser[6164]: auth_db:str_fixup(): Error while querying table version Feb 18 16:31:39 asterix /usr/sbin/ser[6164]: ERROR: fix_expr : fix_actions error Does anyone can help me with this because I stuck and I can't go any further with my configuration. Please help Andrew Cheers

2 1

[Serusers] Problem with time of call invite in Mysql
by mkumar＠mantragroup.com 20 Feb '06

20 Feb '06

Hi All, I have SER and mysql DB in USA and I am calling from India. For all the calls I am dialling from India the time of call invite is getting recorded 5.5 hours ahead the local time in USA. Suppose if time in USA is 6:30 and if I call from India, in mysql DB it is getting stored that I called at 12:00. So for billing if I calculate calls between NOW() and NOW()-30 days, all the calls in this 5.5 hours are missing. Please tell me where I have to change my configuration. I have ser -0.9.4 and Mysql 4.0 on Linux. Thanks for your help and time, Manoj.

1 0

[Serusers] ENUM
by Anderson Alves Albuquerque 20 Feb '06

20 Feb '06

Does someone has a example with ENUM in SER? I need to use SER with ENUM, but I think that the manaul in IPTEL is very sample and it is not complete.

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