- sr-users - kamailio.org

[Serusers] How to configure pa?
by Steven Poveda 09 Jun '05

09 Jun '05

2 1

[Serusers] Re: SEMS not working, problem solved, thank you
by Llanos Serna García-Conde 09 Jun '05

09 Jun '05

1 0

[Serusers] ATA phone not working
by Min ZOU 09 Jun '05

09 Jun '05

Hi, I am trying to get an ATA phone working for SER server. Now the ATA phone can receive calls from softphone(kphone) on a PC, while it cannot call out. (softphone, ATA phone and SER server are in the same subnet) $ serctl ul show 100 <sip:100@192.168.1.100:5600>;q=0.00;expires=802 Any idea on how to resolve the problem? Thanks! -- Best regards, Min

2 2

[Serusers] add_rcv_param() for INVITEs
by Martin Koenig 09 Jun '05

09 Jun '05

Hello, is it possible to use add_rcv_param() also for Contact-HF of INVITEs, instead of fix_nated_contact()? This would make sure that the NATed End Device gets fed exactly the contact as R-URI that it announced in the INVITE: Example R-URI in Loose-Route Requests for fix_nated_contact(): Ser (1.2.3.4): INVITE device(a)4.5.6.7 Route: 1.2.3.4;lr=on Device (10.0.0.1, NAT 4.5.6.7): INVITE device(a)123.123.123.123 With add_rcv_param(): Ser: INVITE device(a)10.0.0.1;rcv_param=4.5.6.7 Route: 1.2.3.4;lr=on Device: INVITE device(a)10.0.0.1 The problem is that the End-Device, in the fix_nated_contact() callflow, has to accept a Request that is not really for this device (IP in Request-URI != IP of the Device). Imagine a Ser checkip uri == myself. This would fail without knowing the public IP of the NAT and hard-conding it into Ser.cfg. Does Ser support the rcv_param in Request-URI also? What do you think? Any experience on this? Regards, Martin

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[Serusers] SER, help please!
by Ryan Pagquil 09 Jun '05

09 Jun '05

Hi, Whenever I uncomment i change the integer value of modparam("usrloc", "db_mode", 0) changed to 1 or 2 and when i about to restart SER using ./serctl stop then ./serctl start, SER strat but when i check ps aux | grep ser no SER is running. I checked /var/log/debug/ and i got this messages: Jun 9 00:45:10 sip ./ser[7553]: DEBUG: tm_shutdown : start Jun 9 00:45:10 sip ./ser[7553]: DEBUG: tm_shutdown : empting DELETE list Jun 9 00:45:10 sip ./ser[7553]: DEBUG: tm_shutdown : empting hash table Jun 9 00:45:10 sip ./ser[7553]: DEBUG: tm_shutdown: releasing timers Jun 9 00:45:10 sip ./ser[7553]: DEBUG: tm_shutdown : removing semaphores Jun 9 00:45:10 sip ./ser[7553]: DEBUG: tm_shutdown : done Jun 9 00:45:10 sip ./ser[7553]: shm_mem_destroy Jun 9 00:45:10 sip ./ser[7553]: destroying the shared memory lock Whats the meaning of destroying the shared memory lock? I'm running SER on FreeBSD/sparc64. Thanks, -- Ryan Pagquil Infodyne Inc. - PhilOnline.com 3603 Antel Global Corporate Center Doña Julia Vargas Ave. Ortigas Center Pasig City Tel: 687-0715 Web: www.philonline.com

2 1

[Serusers] Re: [Sems] SEMS not working
by Llanos Serna García-Conde 09 Jun '05

09 Jun '05

1 0

Re: [Serdev] Re: [Serusers] how to increment cseq? (for uac auth)
by Samuel Osorio Calvo 09 Jun '05

09 Jun '05

Just my 2 cents.... Since SER is transaction aware and not dialog aware, it is not easy to maintain consistent Cseq values during a dialog. I'm sure UAC maintainers are thinking on it and they will come with a solution, but just let them work without stress ;) However, don't think the UAC approach is the right one having TLS around......I would prefer pushing for TLS instead of adding a table of existing dialog in SER with the appropriate values. I'm sure there's lots of providers reading this list so if everybody starts asking for TLS....well, big providers will start providing it for auth. I'm not an expert but I've been told implementing TLS is not a hard job....;) Samuel. Unclassified. >>> "Greger V. Teigre" <greger(a)teigre.com> 06/09/05 07:18AM >>> Michael Ulitskiy wrote: > Greg, do you really believe that avoiding added, but optional > complexity > is good enough reason for not having IMHO necessary functionality? Not at all. My point was that it is extremely important to add functionality where it belongs logically and where you can ensure that regular ser.cfg users are able to use the functionality without understanding the specs. > Again, IMHO, having something better than ip auth is a must in > today's internet. Again, I agree. > On the subject: > I guess the reason is, as I said before, that not only cseq in > original invite must be incremented, but cseq in all subsequent > in-dialog messages must > be adjusted (decremented for messages relayed to callee and > incremented > for messages relayed to called party). That is what, I guess, is hard > to do in today's ser. > I've posted my thoughts on how it possibly could be accomplished > at http://lists.iptel.org/pipermail/serdev/2005-May/004589.html and > haven't received any feedback or comments. > I guess either there's a lack of interest for getting it to work > which I guess is really strange or it's too hard to implement at > current stage. > As always any comments are welcome :) I don't think it's a lack of interest or lack of need for it. All the developers are bombarded with requests and have long lists of core things to do. Sometimes its even hard to get them to evaluate and include patches in CVS. IMHO, increment cseq belongs in the UAC module (can do far too much harm if users play around with it). If you cannot get the maintainers attention, there is only one way: Get your hands dirty ;-) g-) _______________________________________________ Serdev mailing list serdev(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serdev

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[Serusers] Re: [Sems] SEMS not working
by Samuel Osorio Calvo 09 Jun '05

09 Jun '05

First of all: there's no need to send the e-mail more than once...it will just make people not answering. Probably what happens is that you have the old libraries (the ones you compiled with the HEAD version of SEMS) and SEMS rel_0.8.12. Be sure the libraries are correct. Samuel Unclassified. >>> Llanos Serna García-Conde <llanosserna(a)hotmail.com> 06/09/05 09:49AM >>> Hi all, I'm trying to make SEMS work. First I was working with SER 0.8.14 and SEMS cvs but I always got the error consume_request(): consumed. I only want SEMS for voicemail, so the list recommended that I downloaded the SEMS rel_0.8.12, which is supposed to be compatible with SER 0.8.14, but when loading the plug-ins I get errors about undefined symbols. I've been reading previous posts and it seems that it's either an incompatibility between versions, or that SEMS is working with another version's modules or something like that. I don't know how to solve this. Here are the errors I'm getting: Configuration: configuration file: /usr/local/etc/sems/sems12.conf Ser's FIFO: /tmp/ser_fifo our FIFO: /tmp/am_fifo plug-in path: /usr/local/src/sems_rel_0_8_12/lib daemon mode: 0 local IP: 172.25.97.19 (4851) DEBUG: main (sems.cpp:386): Loading application plug-ins (4851) DEBUG: load (AmPlugIn.cpp:89): loading /usr/local/src/sems_rel_0_8_12/lib/apps/bot.so ... (4851) ERROR: loadPlugIn (AmPlugIn.cpp:103): AmPlugIn::loadPlugIn: /usr/local/src/sems_rel_0_8_12/lib/apps/bot.so: undefined symbol: _ZN12AmRequestUAC7dialoutERKSsS1_S1_S1_ (4851) ERROR: load (AmPlugIn.cpp:91): while loading plug-in '/usr/local/src/sems_rel_0_8_12/lib/apps/bot.so' (4850) WARNING: sig_usr_fifo (sems.cpp:84): signal 17 received (4850) INFO: sig_usr_fifo (sems.cpp:93): finished Please help, thanks a lot, Llanos

1 0

[Serusers] fifo issues with ser cvs head version
by Cameron Beattie 09 Jun '05

09 Jun '05

I have installed the cvs head version of ser as of today. Ser seems to start OK but serctl moni reports: Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config I am using the same ser.cfg file as was running successfully on v 0.9.0. Could anyone advise what I'm doing wrong? ser.cfg debug=3 fork=yes log_stderror=yes dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://user:pword@localhost/ser" sock_mode=0666 listen=ip.ip.ip.ip port=5060 children=4 Regards Cameron

1 0

[Serusers] parse_hfs(): Error while parsing headers
by mma 09 Jun '05

09 Jun '05

i 'm connecting to SER, which running on 192.168.1.70 'sip.server.net', using Windows Messenger i add user2 to my contact list i have this errors in response to a SUBSCRIBE request parse_hfs(): Error while parsing headers handle_subscription(): Error while parsing message header can somebody help me Regards //////////////////////////////////////////////////////////////////////////////// 0(2945) SIP Request: 0(2945) method: <REGISTER> 0(2945) uri: <sip:sip.server.net> 0(2945) version: <SIP/2.0> 0(2945) parse_headers: flags=2 0(2945) end of header reached, state=5 0(2945) parse_headers: Via found, flags=2 0(2945) parse_headers: this is the first via 0(2945) After parse_msg... 0(2945) preparing to run routing scripts... 0(2945) parse_headers: flags=100 0(2945) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2945) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2945) parse_headers: flags=200 0(2945) end of header reached, state=9 0(2945) DEBUG: get_hdr_field: <To> [34]; uri=[sip:user1@sip.server.net] 0(2945) DEBUG: to body [<sip:user1@sip.server.net>^M ] 0(2945) get_hdr_field: cseq <CSeq>: <1> <REGISTER> 0(2945) DEBUG: get_hdr_body : content_length=0 0(2945) found end of header 0(2945) find_first_route: No Route headers found 0(2945) loose_route: There is no Route HF 0(2945) grep_sock_info - checking if host==us: 18==18 && [sip.server.net] == [sip.server.net] 0(2945) grep_sock_info - checking if port 5060 matches port 5060 0(2945) parse_headers: flags=ffffffffffffffff 0(2945) parse_headers: flags=4000000 0(2945) build_contact(): Created Contact HF: Contact: <sip:192.168.1.118:14310>;expires=3600^M 0(2945) parse_headers: flags=ffffffffffffffff 0(2945) check_via_address(192.168.1.118, 192.168.1.118, 0) 0(2945) DEBUG:destroy_avp_list: destroying list (nil) 0(2945) receive_msg: cleaning up 0(2945) SIP Request: 0(2945) method: <SUBSCRIBE> 0(2945) uri: <sip:user2@sip.server.net> 0(2945) version: <SIP/2.0> 0(2945) parse_headers: flags=2 0(2945) end of header reached, state=5 0(2945) parse_headers: Via found, flags=2 0(2945) parse_headers: this is the first via 0(2945) After parse_msg... 0(2945) preparing to run routing scripts... 0(2945) parse_headers: flags=100 0(2945) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(2945) DBG:maxfwd:process_maxfwd_header: value 70 decreased to 16 0(2945) parse_headers: flags=10 0(2945) DEBUG: add_param: tag=963f17727b404c5ca92bd1ad97491e13 0(2945) DEBUG: add_param: epid=db4638b430 0(2945) end of header reached, state=29 0(2945) parse_headers: flags=200 0(2945) end of header reached, state=9 0(2945) DEBUG: get_hdr_field: <To> [33]; uri=[sip:user2@sip.server.net] 0(2945) DEBUG: to body [<sip:user2@sip.server.net>^M ] 0(2945) get_hdr_field: cseq <CSeq>: <1> <SUBSCRIBE> 0(2945) DEBUG: get_hdr_body : content_length=0 0(2945) found end of header 0(2945) find_first_route: No Route headers found 0(2945) loose_route: There is no Route HF 0(2945) grep_sock_info - checking if host==us: 18==18 && [sip.server.net] == [sip.server.net] 0(2945) grep_sock_info - checking if port 5060 matches port 5060 0(2945) DEBUG: t_newtran: msg id=2 , global msg id=0 , T on entrance=0xffffffff 0(2945) parse_headers: flags=ffffffffffffffff 0(2945) parse_headers: flags=78 0(2945) t_lookup_request: start searching: hash=53801, isACK=0 0(2945) DEBUG: proceeding to pre-RFC3261 transaction matching 0(2945) DEBUG: t_lookup_request: no transaction found 0(2945) handle_subscription() entered 0(2945) parse_headers: flags=304010 0(2945) parse_hfs(): Error while parsing headers 0(2945) handle_subscription(): Error while parsing message header 0(2945) handle_subscription about to send_reply and return -2 0(2945) DEBUG: t_check: msg id=2 global id=2 T start=0x422efda0 0(2945) DEBUG: t_check: T already found! 0(2945) parse_headers: flags=ffffffffffffffff 0(2945) check_via_address(192.168.1.118, 192.168.1.118, 0) 0(2945) DEBUG: cleanup_uac_timers: RETR/FR timers reset 0(2945) DEBUG: add_to_tail_of_timer[2]: 0x422efde8 0(2945) DEBUG: reply sent out. buf=0x8138d64: SIP/2.0 4..., shmem=0x422f13d8: SIP/2.0 4 0(2945) DEBUG: _reply_light: finished 0(2945) DEBUG:destroy_avp_list: destroying list (nil) 0(2945) receive_msg: cleaning up 0(2945) SIP Request:

1 1

[Serusers] Problem in downloading ser-0.9.0_src.tar.gz
by Gaurav Kansal 09 Jun '05

09 Jun '05

Hi On trying to download ser-0.9.0_src.tar.gz from <ftp://ftp.berlios.de/pub/ser/0.9.0/src/ser-0.9.0_src.tar.gz> ftp://ftp.berlios.de/pub/ser/0.9.0/src/ser-0.9.0_src.tar.gz, i get a pop up window for username/password. In that I give the username/password with which I am registered at iptel.org. But I get an error message and download fails. Can anybody tell me how to download it. Regards, Gaurav Kansal

2 1

[Serusers] SEMS not working
by Llanos Serna García-Conde 09 Jun '05

09 Jun '05

1 0

re:[Serusers] two PSTN gateways with SER
by Ahmad Cheikh Moussa 09 Jun '05

09 Jun '05

Hi! I found this thread in the mail archive and I have a question to this thread. How many pstn gateways can I handle with this failover statement ? Is there a timeout I have to keep in mind, or can I put gateways as many as I have ??? For example. I have one SER Server and 10 PSTN gateways and on every gateway, 4 PRIs. Can I configure the ser server so that the server ask one after another until he found a gateway which accept his call ? The last question is, which isdn features can I use with SER ? I mean can a Ser user make auto hold or call park or something like that ? Thanks in advance, Ahmad >You could not, because ser does not know which PSTN gateway is less >loaded. What you can do is to forward all calls to one of the gateways >and when the gateway refuses the call then you can try the other one. >This can be done using failure_route statement in the configuration >file. > > Jan. > >On 27-05 15:34, Gustavo García Bernardo wrote: >> Hi, >> >> I have a platform based in SER SIP proxy, and i would like to forward >PSTN >> calls to two Cisco gateways. How could implement the logic for send >the >> calls to the less loaded gateway? >> >> Best regards. >> >> G. -- Ahmad Cheikh-Moussa NetUSE AG Dr.-Hell-Straße, 24107 Kiel, Germany Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499 Service: Service(a)NetUSE.DE -- http://NetUSE.DE/

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[Serusers] Ser Presence Server + Windows MSN 4.7
by Simith Nambiar 09 Jun '05

09 Jun '05

Hello Serusers, I'am new to the List , please can you give me the below details , 1.) Does the Presence server in Ser do SUBSCRIBE/ NOTIFY processing and is it compatible with Windows Messenger 4.7 ? what i mean to say is that , does it reflect the current status of the logged in users, and generate appropriate Notifications ? Please let me know ! Thank You, Cheers. Simith

1 0

[Serusers] Ser Presence Server and Windows Messenger
by Simith Nambiar 09 Jun '05

09 Jun '05

Hi All, Does the Presence server in Ser do SUBSCRIBE/ NOTIFY processing and is it compatible with Windows Messenger 4.7 ? what i mean to say is that , does it reflect the current status of the logged in users, and generate appropriate Notifications ? Thank You, Cheers. Simith

1 0

[Serusers] SerWeb ul_add: flags expected
by Mr Greg Plater 09 Jun '05

09 Jun '05

I found the error was an extra field in the table subscriber that was not in pending.. causing mismatch. Pays to check everything when you start to customize serweb.

1 0

[Serusers] SerWeb ul_add: flags expected
by Mr Greg Plater 09 Jun '05

09 Jun '05

Hi When I register a new user I get an error after clicking on the confirmation link in the email I receive..NOTE not before. Also when I have a user in serweb and I try to add an Alias I see this in red: 400 ul_add: flags expected in debug I see : 1(1880) ERROR: ul_add: flags expected This was all working until I decided to make a better ser.cfg file. I think maybe an issue with ser.cfg and needing something for aliases. Because with the old ser.cfg users could register with out issue and I could create an alias. Perms has been added to subscriber and pending tables This has been added to config.php var $flags; $this->flags="128";

1 0

[Serusers] Sems plugin:bot.so, undefined symbol
by Llanos Serna García-Conde 09 Jun '05

09 Jun '05

1 0

[Serusers] optimizing mysql queries
by Iqbal 08 Jun '05

08 Jun '05

Hi The two queries below being used to pull acc info out, seem very slow, are there any Mysql whizzes on the list, the db is optimized, but the second query is just far too long for me to debug. select count(*) from acc t1 right outer join acc t2 on t1.sip_callid=t2.sip_callid and ((t1.totag=t2.totag and t1.fromtag=t2.fromtag) or (t1.totag=t2.fromtag and t1.fromtag=t2.totag)) and t1.sip_method='INVITE' where t2.sip_method='BYE' and isnull(t1.username) and (t2.username='040633' and t2.domain='sip.domain.com') and t2.callee_deleted != '1' (select t1.to_uri as inv_to_uri, t1.sip_to as inv_sip_to,t1.sip_callid as inv_callid, t1.time as inv_time, t1.fromtag as inv_fromtag, t1.sip_status as inv_status, t2.to_uri as bye_to_uri,t2.sip_to as bye_sip_to, t2.sip_callid as bye_callid,t2.time as bye_time,t2.fromtag as bye_fromtag,t2.totag as bye_totag,t2.from_uri as bye_from_uri,t2.sip_from as bye_sip_from,sec_to_time(unix_timestamp(t2.time)-unix_timestamp(t1.time)) as length,ifnull(t1.time, t2.time) as ttime , 'outgoing' as call_type from acc t1 left outer join acc t2 on t1.sip_callid=t2.sip_callid and ((t1.totag=t2.totag and t1.fromtag=t2.fromtag) or (t1.totag=t2.fromtag and t1.fromtag=t2.totag)) and t2.sip_method='BYE' where t1.sip_method='INVITE' and (t1.username='040633' and t1.domain='sip.domain.com') and t1.caller_deleted != '1' ) union (select t1.to_uri as inv_to_uri, t1.sip_to as inv_sip_to,t1.sip_callid as inv_callid,t1.time as inv_time, t1.fromtag as inv_fromtag,t1.sip_status as inv_status,t2.to_uri as bye_to_uri,t2.sip_to as bye_sip_to,t2.sip_callid as bye_callid, t2.time as bye_time,t2.fromtag as bye_fromtag,t2.totag as bye_totag,t2.from_uri as bye_from_uri, t2.sip_from as bye_sip_from,sec_to_time(unix_timestamp(t2.time)-unix_timestamp(t1.time)) as length, ifnull(t1.time, t2.time) as ttime , 'outgoing' as call_type from acc t1 right outer join acc t2 on t1.sip_callid=t2.sip_callid and ((t1.totag=t2.totag and t1.fromtag=t2.fromtag) or (t1.totag=t2.fromtag and t1.fromtag=t2.totag)) and t1.sip_method='INVITE' where t2.sip_method='BYE' and isnull(t1.username) and (t2.username='040633' and t2.domain='sip.domain.com') and t2.callee_deleted != '1' )order by ttime desc limit 0, 20 Now I did a "explain" on these (after having added indexes, since the default install doesn't add indexes. +-------+-------+---------------------+------------+---------+-------+------+--------------------------+ | table | type | possible_keys | key | key_len | ref | rows | Extra | +-------+-------+---------------------+------------+---------+-------+------+--------------------------+ | t1 | ref | username,sip_method | username | 64 | const | 1061 | Using where | | t2 | ref | sip_method | sip_method | 129 | const | 4052 | | | t2 | ref | username,sip_method | username | 64 | const | 1061 | Using where | | t1 | ref | sip_method | sip_method | 129 | const | 2782 | Using where; Not exists | | t1 | index | NULL | username | 64 | NULL | 6837 | Using index | | t2 | ref | sip_method | sip_method | 129 | const | 4052 | Using where; Using index | +-------+-------+---------------------+------------+---------+-------+------+--------------------------+ this for the second query, so rough maths tells me that 351554245281666581856 rows must be examined in order to return the data set..which is alot of rows, in any language. the explain for the first query returns +-------+------+---------------------+------------+---------+-------+------+-------------+ | table | type | possible_keys | key | key_len | ref | rows | Extra | +-------+------+---------------------+------------+---------+-------+------+-------------+ | t1 | ref | username,sip_method | username | 64 | const | 1061 | Using where | | t2 | ref | sip_method | sip_method | 129 | const | 4052 | | +-------+------+---------------------+------------+---------+-------+------+-------------+ which is about 4million. Iqbal

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[Serusers] job opening in NYC
by Igor Balk 08 Jun '05

08 Jun '05

Senior Systems Engineer StanaPhone Communications, LLC is one of the leading Voice over Internet Protocol (VoIP) service providers and is a subsidiary of Intermedia.NET. We are opening a search for a Senior Systems Engineer. The position is located in our headquarters in New York City. With industry leaders as founders and a dynamic and growing team, this is a great opportunity to join an early-stage company with a bright future. If you're an enthusiastic and creative individual with more then 3 years of experience then this is a great career opportunity for you. You must be able to work in an very fast-paced and dynamic environment. As an early stage team member you'll have a unique opportunity to experience an open and collaborative environment in which you'll be able to help shape company strategy and direction. Responsibilities: You will work as a member of the network team, designing and developing next generation VoIP services. You must have strong knowledge of new and existing telephony protocols and their internetworking and a forward-looking perspective with the vision to develop new, innovative services with evolving protocols. Desired Skills: * Demonstrated expertise in the following areas: SIP, ENUM, TLS, TRIP, SS7, development in a Linux / UNIX environment, C and Perl programming, network programming (TCP/UDP) * Working knowledge of the following software systems: Asterisk PBX; SIP Express Router or other SIP Proxy Servers, MySQL and Oracle databases. * Cisco certification is a definite plus. * Strong troubleshooting, debugging, problem solving, and analytical skills * Effective communication (verbal and written) and interpersonal skills * You should be authorized to work in the US as we do not sponsor visas at this time. Intermedia.NET, the parent company, is a leading provider of advanced Web and Exchange Hosting Solutions. The company has been a pioneer in hosting services since 1995 and has continued to innovate, providing its customers with HostPilot, an advanced, industry-recognized control panel for server management and administration. The company responds to customers' most essential needs by providing cost effective, secure and reliable hosting solutions to more than 15,000 businesses. Headquartered in New York City, with offices and resellers worldwide, Intermedia.NET continues to be a customer-focused, financially stable and dependable company. Intermedia.NET is a Microsoft Certified Partner. More information can be found at: www.intermedia.net. Intermedia.NET is an equal opportunity employer and offers competitive wages. We provide top benefits and compensation. Please email resumes with wage history and requirements to jobs(a)stanaphone.com with "Systems Engineer" in the subject line. Please respect our time - no calls or walk-ins.

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[Serusers] ser is failing after a while with sems
by Roger Lewau 08 Jun '05

08 Jun '05

I have set up ser with mediaproxy and sems according to example configs. When I start up mediaproxy, ser and sems all phones register properly and calls can be placed internally (have no external gate yet). I also tried calling a UA not onlie to get sems to take the call and it works great. So I tell my beta tester to start testing.... 1 day after, all calls are rejected because of server error. I have to restart everything and it works again, only to fail the day after... These entries are found in the log for every call I try to make when ser stopped working. Jun 8 22:42:33 ns2 ser[523]: ERROR: t_newtran: transaction already in process 0x29830c58 Jun 8 22:42:33 ns2 ser[523]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) Error message on SJPhone is "number not available" The error message about t_newtran tells me this is caused by a part of the config added for sems to work. Also it seems that after putting the t_newtran in the config there should be some t_release as well since I get this message repeatedly after restarting ser. However I can not find any documentation stating this is necessary or where this should be put in place within ser.cfg. Jun 8 22:53:34 ns2 ser[4898]: WARNING: script writer didn't release transaction If I comment out the SEMS configuration lines withing ser.cfg and restart ser. It works properly again.... First, is there any any one with a ser.cfg script that works with sems without the error/waring messages above being logged? If so please send it to me. Secondly, have any one written any official documentation about this product, including different scenarios? Since the system is widely used I do not understand why there is no official documentation available. Dont say onsip, no more help to find there.

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[Serusers] REGISTER, auth & INVITE
by Lol Zimmerli 08 Jun '05

08 Jun '05

Hello list! I do have something like this in my ser.cfg: if (method=="REGISTER") { if (!www_authorize("domain.com", "subscriber")) { www_challenge("domain.com", "1"); break; }; break; }; It works perfectly. And if I set a line in my snom with inexistant account, of course REGISTER fails: [2]8/6/2005 16:29:39: Registrar 9898(a)212.xxx.xxx.xxx refused with code 401 But, with the same un-REGISTERed phone, I can phone: INVITE works :( I then tried something like this: if (method=="INVITE"){ if (src_ip != "212.xxx.xxx.xxx"){ if (!www_authorize("domain.com", "subscriber")) { www_challenge("domain.com", "1"); break; }; }; else{ route(1); }; }; And then, SER fail to restart: Starting SIP Express Router ERROR: bad config file (6 errors) startproc: exit status of parent of /usr/local/sbin/ser: 255 Is my approach wrong or am I doing a little typo-like error? Best regards -- # Lol Zimmerli // S y s C o ® // http://www.sysco.ch/ Write clearly - don't sacrifice clarity for "efficiency". - The Elements of Programming Style (Kernighan & Plaugher)

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[Serusers] Socketser command interface
by Rodrigo P. Telles 08 Jun '05

08 Jun '05

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Folks, We did a socket interface for serctl command, it means you can run serctl command in your SER box remotely. It's primitive yeat only accept a few commands. It's wrote using Perl Socket interface. Few free to use (at your own risk), change and redistribute! You can download the server and a simple client to test your server from: http://www.devel-it.org/socketser Please read socketser code before use it. Instructions to install, run and use included! We've been using it for 4 months without problems I hope you enjoy! Best regards. - -- ============================================ Rodrigo P. Telles <telles(a)devel.it> TI Manager Devel-IT - http://www.devel.it IVOZ # 1029 +55 14 3324-1200 Bestcom Group ============================================ -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.4 (GNU/Linux) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFCpgz1iLK8unYgEMQRAqRvAJ4nTOaHzCM9H01YTrqXLE+wJ35hEQCeJUWZ +br5k43W1OkYdMRScJiidcQ= =qJvW -----END PGP SIGNATURE-----

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RE: [Serusers] Re-routing to same device in failure_route
by Karun Chemudugunta 08 Jun '05

08 Jun '05

Yes, you are right. Both INVITE messages are same except branch id. Any ideas on how can we solve this issue ? Thanks, karun -----Original Message----- From: ggb(a)tid.es [mailto:ggb@tid.es] Sent: Tuesday, June 07, 2005 3:29 PM To: Karun Chemudugunta; serusers(a)lists.iptel.org Cc: serdev(a)lists.iptel.org Subject: RE: [Serusers] Re-routing to same device in failure_route Hi, It's possible that the Xten softphone detects that the second INVITE is part of a cleared session (the first INVITE rejected with 400 error), and discards it.... G. _____ De: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] En nombre de Karun Chemudugunta Enviado el: martes, 07 de junio de 2005 18:43 Para: serusers(a)lists.iptel.org CC: serdev(a)lists.iptel.org Asunto: [Serusers] Re-routing to same device in failure_route Hi Group, Cisco 5300 -----> SER ----> Xten softphone-A ----> Failed call with error code >400 -------> try to route to Xten softphone A(same as previous) I am trying to route calls to softphones in such way that, if call failed by some reason (>400), trying to route second time to the same soft phone using append_brach() and t_relay() like below. But, soft phone is unable process INVITE message for some reason. But, if I forward failed call different soft phone, call was successful. Am I doing any thing wrong in config ??? Thanks for your help in advance. Regards, Karun ########################################################################################## # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- 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 debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) #dns=no # (cmd. line: -r) #rev_dns=no # (cmd. line: -R) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" #alias=voip.cafe.bevocal.com #alias=66.77.14.238 alias=209.233.189.177 alias=voice.engca.bevocal.com # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/ser/modules/mysql.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/exec.so" loadmodule "/usr/lib/ser/modules/textops.so" #loadmodule "/usr/lib/ser/modules/acc.so" #modparam("acc", "log_level", 2) #modparam("acc", "log_flag", 1) #modparam("acc", "report_cancels", 1) #modparam("acc", "failed_transactions", 1) # 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) # 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 > max_len ) { 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 # record_route(); # loose-route processing if (loose_route()) { t_relay(); 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") { save("location"); break; }; if(method=="INVITE") { log(1,"about lookup in route zero"); if (!lookup("location")) { log(1,"lookup failed :SIP request from config"); sl_send_reply("404", "Not Found"); break; }; t_on_failure("5"); }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } failure_route[5] { log(1,"Inside failure_route five"); revert_uri(); append_branch(); if (!lookup("location")) { log(1,"lookup failed :SIP request from config"); t_reply("404", "Not Found"); break; }; if (!t_relay()) { sl_reply_error(); }; } ######################################################################################

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Re: [Serusers] UAC module: multiple PSTN gateways
by Nour Omar 08 Jun '05

08 Jun '05

Some providers use the IP of the request for authentication and billing. I was wondering how SER can handle that?? Marian Dumitru <marian.dumitru(a)voice-sistem.ro> wrote:Hi Mohammad, UAC module is not present at all in 0.9.0 version, *only* in CVS head version! So far only the CSeq problem is know to us. What exactly doesn't work for you? regards, Marian info(a)beeplove.com wrote: > I tried to configure uac module using this example on ser 0.9.0 > didn't have good luck :-( > > I am just curious to know, > Are you using cvs version of ser to make uac worked? > So far I know, there have a cseq issue. Is it solved yet? > > Thanks > MOhammad > > > Original Message: > ----------------- > From: Marian Dumitru marian.dumitru(a)voice-sistem.ro > Date: Wed, 08 Jun 2005 16:42:24 +0300 > To: info(a)beeplove.com, ext_news(a)appsfarm.com, serusers(a)lists.iptel.org > Subject: Re: [Serusers] UAC module: multiple PSTN gateways > > > Hi Mohammad, > > have a look at an example: > http://www.voice-system.ro/docs/uac/ar01s06.html#ex_auth > > the UAC module is present only in CVS head. > > regards, > Marian > > Mohammad Khan wrote: > >>Did you guys was able to use UAC module successfully? >>If yes, would you please tell what version of ser and uac module you are >>using? >> >>Thanks, >>MOhammad >> >> >>Marian Dumitru wrote: >> >> >>>Hi Cameron, >>> >>>You can specify multiple authentication credentials by just >>>enumerating several "credential" parameters for the module: >>> modparam("uac","credential","user1:realm1:passwd1") >>> modparam("uac","credential","user2:realm2:passwd2") >>> modparam("uac","credential","user3:realm3:passwd3") >>> >>>The module will choose the correct credentials to use based on the >>>"realm" from the auth challenge. >>> >>>regards, >>>Marian >>> >>> >>>Cameron Beattie wrote: >>> >>> >>>>Is it possible to specify multiple authentication credentials within >>>>the UAC module? What I want to do is access multiple PSTN gateways >>>>that each require different credentials. The basic purpose of this is >>>>to allow least cost routing. >>>> >>>>Example >>>>Provider 1 offers cheap calling to +44 (UK) >>>>Provider 2 offers cheap calling to +1 (US) >>>>When user dials a 44 destination ser should relay to Provider 1 >>>>gateway, which will require authentication using Credential 1. >>>>When user dials a 1 destination ser should relay to Provider 2 >>>>gateway, which will require authentication using Credential 2. >>>> >>>>Any advice would be greatfully received. >>>> >>>>Regards >>>> >>>>Cameron >>> >>> >>> >> > -- Voice System http://www.voice-system.ro _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] UAC module: multiple PSTN gateways
by info＠beeplove.com 08 Jun '05

08 Jun '05

I tried to configure uac module using this example on ser 0.9.0 didn't have good luck :-( I am just curious to know, Are you using cvs version of ser to make uac worked? So far I know, there have a cseq issue. Is it solved yet? Thanks MOhammad Original Message: ----------------- From: Marian Dumitru marian.dumitru(a)voice-sistem.ro Date: Wed, 08 Jun 2005 16:42:24 +0300 To: info(a)beeplove.com, ext_news(a)appsfarm.com, serusers(a)iptel.org Subject: Re: [Serusers] UAC module: multiple PSTN gateways Hi Mohammad, have a look at an example: http://www.voice-system.ro/docs/uac/ar01s06.html#ex_auth the UAC module is present only in CVS head. regards, Marian Mohammad Khan wrote: > Did you guys was able to use UAC module successfully? > If yes, would you please tell what version of ser and uac module you are > using? > > Thanks, > MOhammad > > > Marian Dumitru wrote: > >> Hi Cameron, >> >> You can specify multiple authentication credentials by just >> enumerating several "credential" parameters for the module: >> modparam("uac","credential","user1:realm1:passwd1") >> modparam("uac","credential","user2:realm2:passwd2") >> modparam("uac","credential","user3:realm3:passwd3") >> >> The module will choose the correct credentials to use based on the >> "realm" from the auth challenge. >> >> regards, >> Marian >> >> >> Cameron Beattie wrote: >> >>> Is it possible to specify multiple authentication credentials within >>> the UAC module? What I want to do is access multiple PSTN gateways >>> that each require different credentials. The basic purpose of this is >>> to allow least cost routing. >>> >>> Example >>> Provider 1 offers cheap calling to +44 (UK) >>> Provider 2 offers cheap calling to +1 (US) >>> When user dials a 44 destination ser should relay to Provider 1 >>> gateway, which will require authentication using Credential 1. >>> When user dials a 1 destination ser should relay to Provider 2 >>> gateway, which will require authentication using Credential 2. >>> >>> Any advice would be greatfully received. >>> >>> Regards >>> >>> Cameron >> >> >> > > -- Voice System http://www.voice-system.ro -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

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[Serusers] ser+pstn call restrict
by Szabolcs Szasz 08 Jun '05

08 Jun '05

Hi! I am new in ser configuration. My configuration: I use Audiocodes Mediant 2000 for PSTN gateway, ser-0.8.14 installed on my redhat 9.0 box, audiocodes and welltech FXS's as user agents. Now, I can call succesfully sip-->sip, sip-->PSTN and PSTN-->sip. My problem is that I need to restrict some calls for some users, international, national and local (sip-->sip calls). Can anyone help me? Here is my ser.cfg file. Thanks! # # ----------- global configuration parameters ------------------------ debug=4 fork=no log_stderror=yes check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) fifo="/tmp/ser_fifo" listen=10.1.10.10 # ------------------ module loading ---------------------------------- 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/acc.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/uri.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0)# add value to ;lr param to make some broken UAs happy # -- rr params -- modparam("rr", "enable_full_lr", 1) # ------------- accounting parameters modparam("acc", "log_missed_flag", 3) modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) # ------------- usrloc parameters # 2 enables write-back to persistent mysql storage for speed # disable=0, write-through=1 modparam("usrloc", "db_mode", 2) # minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) # database location modparam("usrloc", "db_url", "sql://ser:heslo@localhost/ser") # ------------- auth parameters # database location modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser") # allows clear text passwords in the mysql database modparam("auth_db", "calculate_ha1", yes) # name of password column in mysql database modparam("auth_db", "password_column", "password") # ------------------------- 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 > max_len ) { 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # forward all calls to audiocodes with 0 as the digit header if ( (uri=~"sip:0[0-9]*@*") | (uri=~"sip:1111@*") ){ rewritehostport("10.1.10.1:5060"); forward(10.1.10.1, 5060); 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("10.1.10.10", "subscriber")) { www_challenge("10.1.10.10", "0"); break; }; save("location"); break; }; lookup("aliases"); # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } _____________________________ Szabolcs Szasz Digi Communication SRL H2/3 Petofi Sandor St. Sovata, Mures, Romania sszasz(a)digicomm.ro www.digicomm.ro _____________________________

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[Serusers] Re: [Serdev] ENUM issues - req. for discussion#
by Samuel Osorio Calvo 08 Jun '05

08 Jun '05

Then I see it would take too much effort to add an asynchronous resolver, but I don't think the lack of memory is the right answer: your proposal of using 500 children is more memory demanding than forking, isn't it? You can just add a configuration variable, max_fork=500 and with , for instance, children=3 your deployment will require much less memory than just children=500!!!!!!!!! Nevertheless, for simplicity (in the sense of less points of failure and a more stable system) it is better to stay with the "common" resolver. Samuel. Unclassified. >>> Andrei Pelinescu-Onciul <andrei(a)iptel.org> 06/07/05 04:49PM >>> On Jun 07, 2005 at 14:56, Samuel Osorio Calvo <samuel.osorio@nl.thalesgroup..com> wrote: > >If DNS is slow, or misconfigured (e.g. a zone is delegated to a > >nameserver which is down), the thread will be blocked for several > >seconds. E.g. if you use debian woody and 2 nameservers in > >/etc/resolv.conf, the timeout is 20 seconds. If you are lucky, the OS > >allows configuration of the DNS timeouts. Nevertheless, you have to > >consider that a ser thread will be blocked up to 20 seconds. This has > >impacts on your configuration: > > I don't know the details but would it be really difficult to use an asyncrhonous resolver, such as resiprocate SIP stack does with ARES?? Besides exec_* calls, the main SER's performance bottelneck is the DNS resolving step thus it would be a great improvement adding asyncronous DNS queries. Using asynchronous dns would work as long as you have memory to save the state of the pending dns request. It could be easily attacked in the same way (lots of DNS requests that will take a long time to resolve => out of memory => no more messages processed). Besides using it would mean saving the complete state of the message and of the ser processing of the message in the moment the dns request was made. For example if you make a dns request in module foo, function bar() you should be able to continue from exactly the same point in exactly the same state, when you receive the reply. This would mean something equivalent to saving the whole call trace (the whole stack for that matter) and a lot of global variables. The ammount of complexity involved in converting ser to such a model (where such a detailed state is saved that it makes possible resuming processing at a later time) would be huge. I don't think this would be doable in finite time :-) As an alternative one could fork threads (which would save all the information involved except all the global vars.), or new processes (which would save everything). However in this case we would deal with the forking overhead. This can be attacked too (turning ser into a fork bomb). I think it's much better to start ser with lots of children processes (let's say 500, or the maximum acceptable for your machine configuration). So, I don't think async. dns would be a solution. [...] Andrei

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Re: [Serusers] problem with serctl
by Bruno Oliveira 08 Jun '05

08 Jun '05

Hi, my /var/log/syslog print: "Jun 8 09:18:58 uiara ser: WARNING: fix_socket_list: could not rev. resolve 192.168.1.2 Jun 8 09:18:58 uiara ser: WARNING: fix_socket_list: could not rev. resolve 192.168.1.2 Jun 8 09:18:58 uiara sbin/ser[2490]: Maxfwd module- initializing Jun 8 09:23:05 uiara ser: WARNING: fix_socket_list: could not rev. resolve 192.168.1.2 Jun 8 09:23:05 uiara ser: WARNING: fix_socket_list: could not rev. resolve 192.168.1.2 Jun 8 09:23:05 uiara /usr/local/sbin/ser[2534]: Maxfwd module- initializing Jun 8 09:23:27 uiara nagios: Error: Could not re-connect to database server on host '' for status data. I'll keep trying every 60 seconds... Jun 8 09:30:09 uiara ser: WARNING: fix_socket_list: could not rev. resolve 192.168.1.2 Jun 8 09:30:09 uiara ser: WARNING: fix_socket_list: could not rev. resolve 192.168.1.2 Jun 8 09:30:09 uiara /usr/local/sbin/ser[2555]: Maxfwd module- initializing" thanks for your help! On 6/8/05, Felipe Martins <fmartins(a)mundivox.com> wrote: > > Take a Look at your /var/log/syslog, and send the error message to us ....!!! > > > On Wed, 8 Jun 2005 09:27:33 -0300 > Bruno Oliveira <n1gg4s(a)gmail.com> wrote: > > > Hi, > > i have: > > ser 0.9.0 > > mysql 4.0.24 > > debian > > > > when i type command /sbin/serctl start, the serctl print this line in > > pront command: > > > > "uiara:~# /usr/local/sbin/serctl start > > > > Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed" > > > > Anyone know why ? > > > > thanks for your time. > > > > -- > > []s, > > Bruno "Niggas" Oliveira > > Belo Horizonte - MG > > Msn: n1gg4s(a)gmail.com > > Icq: 176314647 > > > > "Todo o nosso descontentamento por aquilo > > que nos falta procede da nossa falta de > > gratidão por aquilo que temos." > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > -- > Felipe Martins > Mundivox Communications > Tecnologia e Projetos > fmartins(a)mundivox.com > > Tel.: +55 +21 +3820 8839 > Cel.: +55 +21 +9823 8602 > Fax.: +55 +21 +3820 8844 > www.mundivox.com > > > -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

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[Serusers] problem with serctl
by Bruno Oliveira 08 Jun '05

08 Jun '05

Hi, i have: ser 0.9.0 mysql 4.0.24 debian when i type command /sbin/serctl start, the serctl print this line in pront command: "uiara:~# /usr/local/sbin/serctl start Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed" Anyone know why ? thanks for your time. -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

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[Serusers] acc.so error
by mohammed irfan 08 Jun '05

08 Jun '05

Dear sir, I am a final year student doing my engineering in computer science, I am doing a project on SIP. I have problems in ser acc.so module. I am having problems using acc.so. It seems that I can't set the following two modparams: modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 2) But I have no problems setting these two modparams and commenting the top 2: modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) i wanted to keep track of missed_calls in the database. so please help me in this matter. eagerly waiting for reply. I have posted this serusers list in iptel.org. but reply from them please help me. eagerly waiting for reply. with regards irfan

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[Serusers] (no subject)
by mohammed irfan 08 Jun '05

08 Jun '05

Dear sir, I am a final year student doing my engineering in computer science, I am doing a project on SIP. I have problems in ser acc.so module. I am having problems using acc.so. It seems that I can't set the following two modparams: modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 2) But I have no problems setting these two modparams and commenting the top 2: modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) i wanted to keep track of missed_calls in the database. so please help me in this matter. eagerly waiting for reply. I have posted this serusers list in iptel.org. but reply from them please help me. eagerly waiting for reply. with regards irfan

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[Serusers] (no subject)
by mohammed irfan 08 Jun '05

08 Jun '05

Dear sir, I am a final year student doing my engineering in computer science, I am doing a project on SIP. I have problems in ser acc.so module. I am having problems using acc.so. It seems that I can't set the following two modparams: modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 2) But I have no problems setting these two modparams and commenting the top 2: modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) i wanted to keep track of missed_calls in the database. so please help me in this matter. eagerly waiting for reply. I have posted this serusers list in iptel.org. but reply from them please help me. eagerly waiting for reply. with regards irfan

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[Serusers] acc.so errors
by mohammed irfan 08 Jun '05

08 Jun '05

I am having problems using acc.so. It seems that I can't set the following two modparams: modparam("acc", "db_flag", 2) modparam("acc", "db_missed_flag", 2) But I have no problems setting these two modparams and commenting the top 2: modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1)

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[Serusers] ser user in database don't exist
by Ryan Pagquil 08 Jun '05

08 Jun '05

Hi, I'm new in SER. I'm setting it up for just two days in FreeBSD in SPARC64. I run ser_mysql.sh create and everything gone fine. Then i tried to create a user using serctl add and i got an error: ERROR 1045 (28000): Access denied for user 'ser'@'localhost' (using password: YES) introducing a new user to the database failed What seems to be the problem here? FYI, i haven't yet configure ser.cfg... Thanks, Ryan

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[Serusers] Re: [Sems] do I need to upgrade to SER 0.9?
by Llanos Serna García-Conde 08 Jun '05

08 Jun '05

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[Serusers] incoming call routing, voicemail on busy/no answer, load balancing, call forwarding & call display
by Chris St Denis 08 Jun '05

08 Jun '05

I am implementing a rather large scale SER server and there are some features I am having trouble figuring out how to implement. 1. Multi-number Incoming call routing. My users register with my using an id number which then needs to be mapped to 0 or more real phone numbers for incoming call routing, and 0 or 1 for outgoing call display. Is there anyway to do something like this using radius? The backend database of my system will be MS SQL server (not my choice) so radius is being used to connect to it. Multiple numbers will use distinctive rings by inserting an alert info header in with append_hf. 2. Voicemail on busy and no answer and no connect to phone. I think busy can be detected by looking for a response code but not I'm quite sure how, and I think I can use fr_timer for can't connect to phone timeout however, how do I detect no answer with a user customizable timeout? I am also not certain how to send it to the asterisk voicemail server at this point. 3. Call forwarding How can I retrieve call forwarding information from a database (preferably using radius) and forward to it? this also leads to the no answer question above. 4. Load balancing I have gotten failover to work with failure_route and t_on_failure however, I want to not just fail over but also load balance between multiple PSTN gateways. Can this be done using SRV records? round-robin DNS? is there some 'random' type command? 5. Call display I need to be able to block call display for incoming and/or outgoing calls. How can this be achieved from within SER? How can I do these things with stable SER? And, anyone know when .9 is planned to be released? Is it combing soon?

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RE: [Serusers] ACK not sent
by Greg Martin 08 Jun '05

08 Jun '05

It is one call - the IP addresses are: the dialled phone (142.178.21.214), SER (142.178.218.210), the application platform (142.178.213.169) and the re-routed called party (142.178.21.182). The proxy of the application platform uses 142.178.213.201 for return messages as well just to confuse things. The cfg works fine if I don't engage the application platform in the call. > -----Original Message----- > From: Greger V. Teigre [mailto:greger@teigre.com] > Sent: Tuesday, June 07, 2005 12:32 AM > To: Greg Martin; serusers(a)lists.iptel.org > Subject: Re: [Serusers] ACK not sent > > Greg, > I'm not capable of following the trace. There are several IPs > in there (142.178.21.214, 142.178.218.210, 142.178.213.169, > 142.178.21.214, 142.178.21.182). Could you download one of > the configs from http://onsip.org/ and test whether you still > have the same problems? > And if so, make sure that you only include one call with the problem. > g-) > > Greg Martin wrote: > > I'm forwarding to a server which returns an invite with changed > > completion data. I've guarded against loops by assuring the > returned > > invite isn't sent to the server again (I'm not sure that what I'm > > doing is compliant though). The correct phone rings and > when I answer > > the dialed phone thinks it is connected. The rung phone never does > > think it's connected though and a packet capture shows no ack sent > > from the dialed phone which I presume is the problem (the > phones are > > cisco 7960/7940). I'd appreciate any help. Sorry about the long > > message but felt the ngrep dump and cfg were both necessary. > > > > > ********************************************************************** > > ************ > > debug=3 > > fork=yes > > log_stderror=no > > dns=no > > rev_dns=no > > fifo="/tmp/ser_fifo" > > fifo_db_url="mysql://ser:heslo@localhost/ser" > > > > 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" > > > > modparam("usrloc", "db_mode", 1) > > modparam("rr", "enable_full_lr", 1) > > > > route { > > > > # > > > ---------------------------------------------------------------------- > > -- > > # Sanity Check Section # > > > ---------------------------------------------------------------------- > > -- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too > > Many Hops"); break; }; > > > > if (msg:len > max_len) { > > sl_send_reply("513", "Message Overflow"); break; }; > > > > > > # > > > ---------------------------------------------------------------------- > > -- > > # Record Route Section # > > > ---------------------------------------------------------------------- > > -- > > if (method != "REGISTER") { > > record_route(); > > }; > > > > # > > > ---------------------------------------------------------------------- > > -- > > # Loose Route Section # > > > ---------------------------------------------------------------------- > > -- > > if (loose_route()) { > > route(1); > > break; > > }; > > > > > > # > > > ---------------------------------------------------------------------- > > -- > > # Call Type Processing Section # > > > ---------------------------------------------------------------------- > > -- > > if (uri != myself) { > > route(1); > > break; > > }; > > > > > > if (method == "REGISTER") { > > route(2); > > break; > > }; > > > > > > if (method == "INVITE") { > > if (!lookup("location")) { > > sl_send_reply("404", "User Not Found"); break; }; > > > > if(src_ip != "142.178.213.169" > > && src_ip != "142.178.213.201" > > && src_ip != "142.178.213.170" > > && src_ip != "142.178.213.202") { > > rewritehost("142.178.213.169"); > > forward(uri:host, uri:port); > > break; > > }; > > } > > > > route(1); > > > > } > > > > route[1] { > > # > > > ---------------------------------------------------------------------- > > -- > > # Default Message Handler # > > > ---------------------------------------------------------------------- > > -- > > > > if (!t_relay()) { > > sl_reply_error(); > > }; > > > > } > > > > route[2] { > > # > > > ---------------------------------------------------------------------- > > -- > > # REGISTER Message Handler # > > > ---------------------------------------------------------------------- > > -- > > if (!save("location")) { > > sl_reply_error(); > > }; > > } > > > > > ********************************************************************** > > ***************** > > > > interface: qfe0 (142.178.218.0/255.255.254.0) > > filter: ip and ( port 5060 ) > > # > > U 142.178.21.214:50090 -> 142.178.218.210:5060 INVITE > > sip:5553331@142.178.218.210 SIP/2.0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: <sip:5553331@142.178.218.210>. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:17 GMT. > > CSeq: 101 INVITE. > > User-Agent: CSCO/4. > > Contact: <sip:5551111@142.178.21.214:5060>. > > Expires: 180. > > Content-Type: application/sdp. > > Content-Length: 249. > > Accept: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. > > s=SIP Call. > > c=IN IP4 142.178.21.214. > > t=0 0. > > m=audio 22102 RTP/AVP 0 8 18 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:18 G729/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.213.169:5060 INVITE > > sip:5553331@142.178.213.169:5060;user=ip SIP/2.0. > > Max-Forwards: 10. > > Record-Route: > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: <sip:5553331@142.178.218.210>. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:17 GMT. > > CSeq: 101 INVITE. > > User-Agent: CSCO/4. > > Contact: <sip:5551111@142.178.21.214:5060>. > > Expires: 180. > > Content-Type: application/sdp. > > Content-Length: 249. > > Accept: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. > > s=SIP Call. > > c=IN IP4 142.178.21.214. > > t=0 0. > > m=audio 22102 RTP/AVP 0 8 18 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:18 G729/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 100 Trying. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > To: <sip:5553331@142.178.218.210>. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 0. > > . > > > > # > > U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 100 Trying. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > To: <sip:5553331@142.178.218.210>. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 0. > > . > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 INVITE > > sip:5552221@142.178.218.210:5060 SIP/2.0. > > Via: SIP/2.0/UDP > > 142.178.213.169:5060;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Max-Forwards: 9. > > Record-Route: <sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>. > > Record-Route: > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: <sip:5553331@142.178.218.210>. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 249. > > Date: Mon, 06 Jun 2005 22:24:17 GMT. > > User-Agent: CSCO/4. > > Contact: <sip:5551111@142.178.21.214:5060>. > > Expires: 180. > > Content-Type: application/sdp. > > Accept: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. > > s=SIP Call. > > c=IN IP4 142.178.21.214. > > t=0 0. > > m=audio 22102 RTP/AVP 0 8 18 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:18 G729/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 100 > trying -- > > your call is important to us. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531 > ;received=142.178.213.201. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > To: <sip:5553331@142.178.218.210>. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Server: Sip EXpress router (0.9.1 (sparc64/solaris)). > > Content-Length: 0. > > Warning: 392 142.178.218.210:5060 "Noisy feedback tells: pid=14006 > > req_src_ip=142.178.213.201 req_src_port=37403 > > in_uri=sip:5552221@142.178.218.210:5060 > > out_uri=sip:5552221@142.178.21.182:5060;user=ip via_cnt==3". . > > > > # > > U 142.178.218.210:5060 -> 142.178.21.182:5060 INVITE > > sip:5552221@142.178.21.182:5060;user=ip SIP/2.0. > > Record-Route: > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKcz > K75.kDwlvStR8IbIbYqg~~531. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Max-Forwards: 8. > > Record-Route: <sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>. > > Record-Route: > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: <sip:5553331@142.178.218.210>. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 249. > > Date: Mon, 06 Jun 2005 22:24:17 GMT. > > User-Agent: CSCO/4. > > Contact: <sip:5551111@142.178.21.214:5060>. > > Expires: 180. > > Content-Type: application/sdp. > > Accept: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. > > s=SIP Call. > > c=IN IP4 142.178.21.214. > > t=0 0. > > m=audio 22102 RTP/AVP 0 8 18 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:8 PCMA/8000. > > a=rtpmap:18 G729/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.21.182:52535 -> 142.178.218.210:5060 SIP/2.0 100 Trying. > > Via: SIP/2.0/UDP > > 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: <sip:5553331@142.178.218.210>. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:23 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Length: 0. > > . > > > > # > > U 142.178.21.182:52536 -> 142.178.218.210:5060 SIP/2.0 180 Ringing. > > Via: SIP/2.0/UDP > > 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:23 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Length: 0. > > . > > > > # > > U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 180 Ringing. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:23 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Length: 0. > > . > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 180 Ringing. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 0. > > Date: Mon, 06 Jun 2005 22:24:23 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > . > > > > # > > U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 180 Ringing. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 0. > > Date: Mon, 06 Jun 2005 22:24:23 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > . > > > > # > > U 142.178.21.182:52537 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.21.182:52538 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.21.182:52539 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.21.182:52540 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.21.182:52541 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP > > > 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwl > > vStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP > > 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > CSeq: 101 INVITE. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > Content-Type: application/sdp. > > Content-Length: 201. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.218.210;branch=0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > Record-Route: > > > <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr > =on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.2 > 10;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > CSeq: 101 INVITE. > > Content-Length: 201. > > Date: Mon, 06 Jun 2005 22:24:26 GMT. > > Server: CSCO/4. > > Contact: <sip:5552221@142.178.21.182:5060>. > > Content-Type: application/sdp. > > . > > v=0. > > o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. > > s=SIP Call. > > c=IN IP4 142.178.21.182. > > t=0 0. > > m=audio 22646 RTP/AVP 0 101. > > a=rtpmap:0 PCMU/8000. > > a=rtpmap:101 telephone-event/8000. > > a=fmtp:101 0-15. > > > > # > > U 142.178.21.214:50096 -> 142.178.218.210:5060 BYE > > > sip:142.178.218.210:5060;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on > > SIP/2.0. > > Via: SIP/2.0/UDP 142.178.21.214:5060. > > From: "Greg Martin" > > <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. > > To: > > <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. > > Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. > > Date: Mon, 06 Jun 2005 22:24:31 GMT. > > CSeq: 102 BYE. > > User-Agent: CSCO/4. > > Content-Length: 0. > > Route: > > > <sip:tc87sv01:5060;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210:5060; > > ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>, > > <sip:5552221@142.178.21.182:5060>. . > > > > # > > > > Greg Martin > > Technology & Operations > > TELUS Communication Inc. > > office: 780-493-2786 > > cell: 780-718-4139 > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > >

2 1

[Serusers] Load balancing
by Chris St Denis 08 Jun '05

08 Jun '05

I have gotten failover to work with failure_route and t_on_failure however, I want to not just fail over but also load balance. Can this be done using SRV records? round-robin DNS? is there some 'random' type command?

2 1

Re: [Serusers] add_rcv_param() for INVITEs
by Klaus Darilion 08 Jun '05

08 Jun '05

Hi Martin! As far as I understand, the problem is not the initial INVITE, but the in-dialog request, which use the fixed contact in the request URI. Correct? Do you have problems with certain clients or is it an academical question? regards, klaus Martin Koenig wrote: > Hello Klaus, > > I know that fix_nated_register works for REGISTERs, the question was > whether add_rcv_param() works for INVITEs ;) > > Answer: It does not. > > Regards, > Martin > > Klaus Darilion wrote: > >> I thin k I solved the problem using fix_nated_register instead of >> fix_nated_contact for REGISTER messages. (using nathelper) >> >> regards, >> klaus >> >> Martin Koenig wrote: >> >>> Hello, >>> >>> is it possible to use add_rcv_param() also for Contact-HF of INVITEs, >>> instead of fix_nated_contact()? >>> >>> This would make sure that the NATed End Device gets fed exactly the >>> contact as R-URI that it announced in the INVITE: >>> >>> Example R-URI in Loose-Route Requests for fix_nated_contact(): >>> >>> Ser (1.2.3.4): >>> INVITE device(a)4.5.6.7 >>> Route: 1.2.3.4;lr=on >>> >>> Device (10.0.0.1, NAT 4.5.6.7): >>> INVITE device(a)123.123.123.123 >>> >>> With add_rcv_param(): >>> >>> Ser: >>> INVITE device(a)10.0.0.1;rcv_param=4.5.6.7 >>> Route: 1.2.3.4;lr=on >>> >>> Device: >>> INVITE device(a)10.0.0.1 >>> >>> The problem is that the End-Device, in the fix_nated_contact() >>> callflow, has to accept a Request that is not really for this device >>> (IP in Request-URI != IP of the Device). Imagine a Ser checkip uri == >>> myself. This would fail without knowing the public IP of the NAT and >>> hard-conding it into Ser.cfg. >>> >>> Does Ser support the rcv_param in Request-URI also? >>> >>> What do you think? Any experience on this? >>> >>> Regards, >>> Martin >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >> > >

1 0

[Serusers] ERROR: t_get_trans_ident: no transaction found
by Roger Lewau 08 Jun '05

08 Jun '05

Whenever I make a call I get these entries in the log.... Jun 7 17:56:29 ns2 ser[520]: WARNING: script writer didn't release transaction Jun 7 17:56:42 ns2 ser[522]: ERROR: t_get_trans_ident: no transaction found Jun 7 17:56:42 ns2 ser[522]: assemble_msg: t_get_trans_ident failed Jun 7 17:56:42 ns2 ser[522]: ERROR:tm:t_write_req: Error int assemble_msg Jun 7 17:56:42 ns2 ser[522]: ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established If the user is online I get connected properly, if user is offline I get connected to sems. So It seems to be working after all. But I had this config running for three days then I got a lot of errors regarding t_newtran and out of memory... and phones lost their registration. Also does any have any idea where to put t_release() to avoid the first WARNING message... This is my ser.cfg debug=3 fork=yes log_stderror=no listen=212.247.91.237 port=5060 children=4 alias=sip.serverhallen.com dns=no rev_dns=no fifo_mode=0666 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" #LADDA IN DE MODULER VI BEHÖVER #--------------------------------------------------------------------------- ------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" #MySQL database support loadmodule "/usr/local/lib/ser/modules/sl.so" #Stateless replies loadmodule "/usr/local/lib/ser/modules/tm.so" #Transaction Management loadmodule "/usr/local/lib/ser/modules/rr.so" #Routing and Record-Routing loadmodule "/usr/local/lib/ser/modules/maxfwd.so" #Max-Forwards check loadmodule "/usr/local/lib/ser/modules/usrloc.so" #User location support loadmodule "/usr/local/lib/ser/modules/registrar.so" #Registrar loadmodule "/usr/local/lib/ser/modules/auth.so" #Common functions for digest authentication loadmodule "/usr/local/lib/ser/modules/auth_db.so" #Database support for digest authentication loadmodule "/usr/local/lib/ser/modules/uri.so" #Various URI checks loadmodule "/usr/local/lib/ser/modules/uri_db.so" #URI checking using DB loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" #RTP Mediaproxy for NATed UAs loadmodule "/usr/local/lib/ser/modules/nathelper.so" #Functions to handle NAT traversal loadmodule "/usr/local/lib/ser/modules/textops.so" #Message Textual Operations loadmodule "/usr/local/lib/ser/modules/avpops.so" #avp operations, pseudo-variable support loadmodule "/usr/local/lib/ser/modules/domain.so" #checks if a host part of an URI is "local" or not. loadmodule "/usr/local/lib/ser/modules/permissions.so" #determine if a call has appropriate permission to be established #VARIABLER FÖR DATABAS OCH HANTERINGA AUTENTISIERING #--------------------------------------------------------------------------- ------------------- modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) #VARIABLER FÖR HANTERING AV NATade UAs #--------------------------------------------------------------------------- -------------------- 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","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("registrar", "nat_flag", 6) #VARIABLER FÖR HANTERING AV PSTN #--------------------------------------------------------------------------- -------------------- modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") #MEDDELANDEHANTERING AV SIP MEDDELANDEN #--------------------------------------------------------------------------- -------------------- #Alla meddelanden kommer behandlas enligt detta regelverk i sekventiell ordning route { # ----------------------------------------------------------------- # Sanity Check Section, Kontrollera hur många gånger ett meddelande tagits emot. # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section, Sektion som lägger till vår egen sip server i routing kedjan så att svaren kommer tillbaka till oss. # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { record_route_preset("212.247.91.237:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section, Denna sektion hanterar nedtagning av samtal # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { # end_media_session(); # if(!t_write_req("/tmp/am_fifo","bye")) { t_reply("500","error contacting sems"); }; }; # ----------------------------------------------------------------- # 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); break; }; # ----------------------------------------------------------------- # Call Type Processing Section, om meddelandet inte skulle skickas vidare är det förmodligen riktat till SER # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { # switch to statefull mode: if (!t_newtran()){ sl_send_reply("500","could not create transaction"); break; }; if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User not found"); break; }; }; log("NO MATCH"); route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler, Standardhantering av Meddelanden # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()){ if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (uri=~"^sip:[+|00][0-9]*@") { route(4); break; }; if (uri=~"^sip:0[1-9]*@") { route(4); break; }; if (uri=~"sip:100.*@") { if(!t_write_req("/tmp/am_fifo","conference")) { t_reply("500","error contacting sems"); }; break; }; if (!lookup("location")) { #Send caller to users answering machine if uri is local if(!t_write_req("/tmp/am_fifo","voicemail")) { t_reply("500","error contacting sems"); }; break; }; if (method=="CANCEL") { route(1); break; }; route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost(" 192.168.0.1 "); avp_write("i:45", "inv_timeout"); route(5); route(1); } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler, aktiverar proxyn # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[6] { # ------------------------------------------------------------------------ # ACK Handler # ------------------------------------------------------------------------ # ------------------------------------------------------------------------ # Aliases Section, hanterar acks # ------------------------------------------------------------------------ lookup("aliases"); if (uri!=myself) { route(1); break; }; lookup("location"); route(1); } 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

RE: [Serusers] Mediaproxy in LAN (?)
by a cairo 08 Jun '05

08 Jun '05

Thanks, but if all clients connect to ser how balancing happen from mediaproxy? -----Original Message----- From: "Shaikat Mahmud" <shaikat(a)stitel.com> To: "'a cairo'" <a.cairo(a)scsitalia.it>, "'Giudice, Salvatore'" <Salvatore.Giudice(a)FMR.COM>, "'bogdan aux'" <aux1d(a)yahoo.com>, <serusers(a)lists.iptel.org> Date: Sat, 4 Jun 2005 16:29:36 +0600 Subject: RE: [Serusers] Mediaproxy in LAN (?) > Hi, > > In mediaproxy, README file has all the architectural information. > > Is mediaproxy between client and ser? > No, its some thing like below, > > > Ua1--------------> SER-------------->UA2 > | > | (Mediaproxy module requesting for a session) > | > | > | > MediaProxy > > > Then RTP flow will be like following: > Ua1 <-------> MediaProxy <-------> Ua2 > > Read the README file for details. > shaikat > > > -----Original Message----- > From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On > Behalf Of a cairo > Sent: Saturday, June 04, 2005 4:14 PM > To: Giudice, Salvatore; bogdan aux; serusers(a)lists.iptel.org > Subject: RE: [Serusers] Mediaproxy in LAN (?) > > Hi all, > i don't understand architecture of mediaproxy with ser ! > Is mediaproxy between client and ser? > Thanks > > -----Original Message----- > From: "Giudice, Salvatore" <Salvatore.Giudice(a)FMR.COM> > To: "bogdan aux" <aux1d(a)yahoo.com>, <serusers(a)lists.iptel.org> > Date: Tue, 17 May 2005 20:10:46 -0400 > Subject: RE: [Serusers] Mediaproxy in LAN (?) > > > I am trying to route a call from a non-routable 10.x.x.x network to a > > phone on the internet using ser 0.9.0 and mediaproxy 1.31. Aka: PSTN > to > > SIP from my asterisk to an x-lite > > > > I found some references indicating that I should call: > > > > > > if ( src_ip = IP_GATEWAY) > > { > > force_rport(); > > fix_contact(); > > use_media_proxy(); > > }; > > > > Supposedly, you need the force_rport and the fix_contact to put media > > proxy in the middle of the two rtp sessions, but I have et to see > this > > actually work. Has anyone been able to pass a call through media > proxy > > from a GW on a non-routable to a routable network, or even between > two > > ip's on the same network as the mediaproxy? > > > > If so, please share your configs. > > > > -----Original Message----- > > From: bogdan aux [mailto:aux1d@yahoo.com] > > Sent: Friday, May 13, 2005 6:16 AM > > To: serusers(a)lists.iptel.org > > Subject: [Serusers] Mediaproxy in LAN (?) > > > > I want to route RTP packets between 2 computers in the > > same network through Mediaproxy. I know this may have > > no sense, but it's just a test configuration. > > The call use_media_proxy() seems to have no effect, > > the RTP packets go directly from one computer to > > another. > > My guess is mediaproxy makes a test on the IPs (they > > are 192.168.147.2 and 192.168.147.3) and refuses to > > proxy the call. > > Is mediaproxy working just between different networks > > or it's an error in my configuration file? > > > > > > > > > > Yahoo! Mail > > Stay connected, organized, and protected. Take the tour: > > http://tour.mail.yahoo.com/mailtour.html > > > > _______________________________________________ > > 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 >

3 2

[Serusers] Re-routing to same device in failure_route
by Karun Chemudugunta 07 Jun '05

07 Jun '05

Hi Group, Cisco 5300 -----> SER ----> Xten softphone-A ----> Failed call with error code >400 -------> try to route to Xten softphone A(same as previous) I am trying to route calls to softphones in such way that, if call failed by some reason (>400), trying to route second time to the same soft phone using append_brach() and t_relay() like below. But, soft phone is unable process INVITE message for some reason. But, if I forward failed call different soft phone, call was successful. Am I doing any thing wrong in config ??? Thanks for your help in advance. Regards, Karun ########################################################################################## # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- 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 debug=7 fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) #dns=no # (cmd. line: -r) #rev_dns=no # (cmd. line: -R) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" #alias=voip.cafe.bevocal.com #alias=66.77.14.238 alias=209.233.189.177 alias=voice.engca.bevocal.com # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/ser/modules/mysql.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/exec.so" loadmodule "/usr/lib/ser/modules/textops.so" #loadmodule "/usr/lib/ser/modules/acc.so" #modparam("acc", "log_level", 2) #modparam("acc", "log_flag", 1) #modparam("acc", "report_cancels", 1) #modparam("acc", "failed_transactions", 1) # 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) # 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 > max_len ) { 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 # record_route(); # loose-route processing if (loose_route()) { t_relay(); 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") { save("location"); break; }; if(method=="INVITE") { log(1,"about lookup in route zero"); if (!lookup("location")) { log(1,"lookup failed :SIP request from config"); sl_send_reply("404", "Not Found"); break; }; t_on_failure("5"); }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } failure_route[5] { log(1,"Inside failure_route five"); revert_uri(); append_branch(); if (!lookup("location")) { log(1,"lookup failed :SIP request from config"); t_reply("404", "Not Found"); break; }; if (!t_relay()) { sl_reply_error(); }; } ######################################################################################

2 1

[Serusers] Re: [Serdev] ENUM issues - req. for discussion
by Samuel Osorio Calvo 07 Jun '05

07 Jun '05

Hi Klaus, responses inline...(command and file names are taken from my memory, so don't pay much attention at the spelling) Samuel. Unclassified. >>> Klaus Darilion <klaus.mailinglists(a)pernau.at> 06/07/05 01:43PM >>> >Hi all! > >Some time of using ser with ENUM revealed several problems which I would >like to dicuss with you. Be aware - this email is long! > >ENUM is a wonderful thing for call routing nevertheless, as it is DNS >based there are some important things an ENUM aware application has to >consider: > 1. You never know how long the lookup takes > 2. You never know if the lookup will fail or succeed > 3. If the lookup was successful, you never may trust the result> > >I will explain this points in detail know: > >1+2: >Using enum, the application is giving control to the DNS resolver of the >OS and the DNS infrastructure. Thus, the ser thread which performs the >ENUM lookup will be blocked until there is a result from the system's >DNS resolver. > >If DNS is slow, or misconfigured (e.g. a zone is delegated to a >nameserver which is down), the thread will be blocked for several >seconds. E.g. if you use debian woody and 2 nameservers in >/etc/resolv.conf, the timeout is 20 seconds. If you are lucky, the OS >allows configuration of the DNS timeouts. Nevertheless, you have to >consider that a ser thread will be blocked up to 20 seconds. This has >impacts on your configuration: I don't know the details but would it be really difficult to use an asyncrhonous resolver, such as resiprocate SIP stack does with ARES?? Besides exec_* calls, the main SER's performance bottelneck is the DNS resolving step thus it would be a great improvement adding asyncronous DNS queries. >Typically, you use some kind of the following logic: >if (uri =~ "+[0-9].") { > if (enum_lookup()) { > t_relay(); break; > } else { > forward to PSTN gateway; break; > } >} > >Thus, the INVITE will be received and the ENUM lookup will be performed. >If the lookup will take longer than 0.5s, the SIP client will start >restransmitting the INVITE. Thus, another thread will process this >INVITE and enother ENUM lookup will be performed. After several seconds, >all of ser's threads will be blocked with ENUM lookups and your SIP >proxy will not handle any requests until the DNS query times out. Thus, >it is very easy to generate a DoS attack against the proxy. >Another funny thing is, that the SIP client will detect a proxy error >and hangs up, but the INVITEs are still processed in the SIP proxy and >after the timeout forwarded to the PSTN gateway. > >A solution to stop the retransmission is to immediatle >sl_send_reply("100","Trying"), But this rises another problem. Now, if >the caller hangs up before the DNS timeout, the SIP client will send >CANCEL (as it received 100) to the SIP proxy. But the SIP proxy can not >cancel the transaction, as it is not genereated yet - the INVITE thread >is still waiting for the ENUM lookup and the transaction will be >generated after the ENUM lookup (after 20s timeout). Thus, we still end >up with an INVITE forwarded to the PSTN gateway although the SIP client >already hang up. > >I thought of using t_newtran and t_forward_nonack_uri instead of t_realy >to generate the transaction before doing the ENUM lookup. Thus, the >thread which will process the CANCEL should find a transaction and stop >it. But will this really prevent the INVITE sent to the PSTN gateway, >once the DNS times out? (not tested) I think the best way would be to add a reply(100) AND t_forward_nonack, something like: sl_send_reply("100","Trying"); t_newtran(); uri =~ "+[0-9].") { if (enum_lookup()) { t_forward_nonack(); break;} else{ t_forward_nonac();break; } But it adds a complexity level which will lead to many non-working config files......but I vote for it! > >btw: this problem does not only occour for ENUM lookups, but for all DNS >lookups (SRV, A, AAAA). > >3: >If the ENUM lookup succeeds, you never may trust the result. It may be a >invalid SIP URI, or a tel: URI, or anything else a user puts into its >NAPTRs. This may result in a failed transaction, or like revealed at the >ENUM plugtest in failed accounting. Even worse, maybe it is possible to >complete crash ser using realy bad formated URIs? > >Thus you can't avoid doing some URI checks against the URI received from >the ENUM lookup. Perfomance issues are no valid arguements! Once I give >control to external services (DNS, radius, exec), the perfomance >penalties due to parsing the SIP URI are much more less than due to the >ENUM lookup. > >In case of ser, I would do the URI parsing in the ENUM module, or maybe >generate a dedicated function/module for checking SIP URIs inside the >routing logic. Thus, I can also check the result of exec calls. Since URI check is a really global action, it should be implemented in SER's core, providing commands to the config file (something like uri_check(To|From|Req-URI|Contact) ) and functions to be called from any module. > >I'm eagerly waiting for your opinions. > >regards, >Klaus _______________________________________________ Serdev mailing list serdev(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serdev

3 2

[Serusers] Re: RADIUS failure for long domains returned by ENUM queries
by Adrian Georgescu 07 Jun '05

07 Jun '05

Juha, what if the response from DNS is greater then 253 bytes? Juha Heinanen wrote: >adrian, > >the value of the attribute that you try to add is 202 bytes long. >my /usr/include/radiusclient.h defines > >#define AUTH_STRING_LEN 128 /* maximum of 253 */ > >typedef struct value_pair >{ > char name[NAME_LENGTH + 1]; > int attribute; > int type; > UINT4 lvalue; > char strvalue[AUTH_STRING_LEN + 1]; > struct value_pair *next; >} VALUE_PAIR; > >thus the value is too long and would fail. > >on the other hand, my /usr/include/radiusclient-ng defines: > >#define AUTH_STRING_LEN 253 /* maximum of 253 */ > >which one are you using in your test setup? > >-- juha > >

3 2

[Serusers] ENUM issues - req. for discussion
by Klaus Darilion 07 Jun '05

07 Jun '05

Hi all! Some time of using ser with ENUM revealed several problems which I would like to dicuss with you. Be aware - this email is long! ENUM is a wonderful thing for call routing nevertheless, as it is DNS based there are some important things an ENUM aware application has to consider: 1. You never know how long the lookup takes 2. You never know if the lookup will fail or succeed 3. If the lookup was successful, you never may trust the result I will explain this points in detail know: 1+2: Using enum, the application is giving control to the DNS resolver of the OS and the DNS infrastructure. Thus, the ser thread which performs the ENUM lookup will be blocked until there is a result from the system's DNS resolver. If DNS is slow, or misconfigured (e.g. a zone is delegated to a nameserver which is down), the thread will be blocked for several seconds. E.g. if you use debian woody and 2 nameservers in /etc/resolv.conf, the timeout is 20 seconds. If you are lucky, the OS allows configuration of the DNS timeouts. Nevertheless, you have to consider that a ser thread will be blocked up to 20 seconds. This has impacts on your configuration: Typically, you use some kind of the following logic: if (uri =~ "+[0-9].") { if (enum_lookup()) { t_relay(); break; } else { forward to PSTN gateway; break; } } Thus, the INVITE will be received and the ENUM lookup will be performed. If the lookup will take longer than 0.5s, the SIP client will start restransmitting the INVITE. Thus, another thread will process this INVITE and enother ENUM lookup will be performed. After several seconds, all of ser's threads will be blocked with ENUM lookups and your SIP proxy will not handle any requests until the DNS query times out. Thus, it is very easy to generate a DoS attack against the proxy. Another funny thing is, that the SIP client will detect a proxy error and hangs up, but the INVITEs are still processed in the SIP proxy and after the timeout forwarded to the PSTN gateway. A solution to stop the retransmission is to immediatle sl_send_reply("100","Trying"), But this rises another problem. Now, if the caller hangs up before the DNS timeout, the SIP client will send CANCEL (as it received 100) to the SIP proxy. But the SIP proxy can not cancel the transaction, as it is not genereated yet - the INVITE thread is still waiting for the ENUM lookup and the transaction will be generated after the ENUM lookup (after 20s timeout). Thus, we still end up with an INVITE forwarded to the PSTN gateway although the SIP client already hang up. I thought of using t_newtran and t_forward_nonack_uri instead of t_realy to generate the transaction before doing the ENUM lookup. Thus, the thread which will process the CANCEL should find a transaction and stop it. But will this really prevent the INVITE sent to the PSTN gateway, once the DNS times out? (not tested) btw: this problem does not only occour for ENUM lookups, but for all DNS lookups (SRV, A, AAAA). 3: If the ENUM lookup succeeds, you never may trust the result. It may be a invalid SIP URI, or a tel: URI, or anything else a user puts into its NAPTRs. This may result in a failed transaction, or like revealed at the ENUM plugtest in failed accounting. Even worse, maybe it is possible to complete crash ser using realy bad formated URIs? Thus you can't avoid doing some URI checks against the URI received from the ENUM lookup. Perfomance issues are no valid arguements! Once I give control to external services (DNS, radius, exec), the perfomance penalties due to parsing the SIP URI are much more less than due to the ENUM lookup. In case of ser, I would do the URI parsing in the ENUM module, or maybe generate a dedicated function/module for checking SIP URIs inside the routing logic. Thus, I can also check the result of exec calls. I'm eagerly waiting for your opinions. regards, Klaus

5 14

[Serusers] outbound proxy
by harry gaillac 07 Jun '05

07 Jun '05

Hello ser users, I wish all sip message are handle by ser. phone1 -------------------------- | ser+mediaproxy|------internet phone2 -------------------------- In order to send all sip messages via ser I set outbound sip proxy (ser/public ip) for sip clients. However ser don't route invite message for other domains.Why? Harry # $Id: ser.cfg,v 1.27 2005/03/10 14:16:25 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) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* 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" fifo_db_url="mysql://ser:heslo@localhost/ser" user=root group=ser fifo_user=root # owner of the ser fifo fifo_group=ser fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) listen=80.119.9.14 # # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/mysql.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/uri.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/mediaproxy.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/permissions.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params-- modparam("usrloc", "db_mode", 2) # -- autdb_params-- modparam("auth_db", "calculate_ha1", yes) 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) # -- nathelper params -- modparam("nathelper", "natping_interval", 0) modparam("nathelper", "rtpproxy_disable", 1) # --mediaproxy param -- modparam("mediaproxy", "natping_interval", 30) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") # -- registrar params -- modparam("registrar", "nat_flag", 6) # -- tm params -- modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # --domain params -- modparam("domain", "db_mode", 1) # Use caching # -- permissions params -- modparam("permissions", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") # ------------------------- 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 >= max_len ) { 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(); }; if (method=="CANCEL" || method=="BYE") { end_media_session(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if (has_totag() && method=="INVITE" || method=="ACK") { if (nat_uac_test("3")) { setflag(6); force_rport(); fix_contact(); }; use_media_proxy(); }; route(1); break; }; # Requests to others domains if (!is_from_local()) { # 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 (is_from_local()) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; }; } route[1] { ########################### # Default Message Handler # ########################### t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { end_media_session(); }; sl_reply_error(); }; } route[2] { ############################ # Register Message Handler # ############################ if (nat_uac_test("3")) { setflag(6); force_rport(); fix_contact(); }; if (!www_authorize("nxs.yi.org", "subscriber")) { www_challenge("nxs.yi.org", "0"); sl_send_reply("401", "Unauthorized"); break; }; if (!save("location")) { sl_reply_error(); break; }; } route[3] { ############################# # CANCEL and INVITE handler # ############################# if (nat_uac_test("3")) { setflag(6); }; if (isflagset(6)) { force_rport(); fix_contact(); use_media_proxy(); }; if (!is_from_local()) { route(1); break; }; # SIP destinations from PSTN if (src_ip==192.168.0.50 && allow_trusted()) { route(4); } else if (src_ip==192.168.0.50 && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("403", "Forbidden"); break; }; }; # SIP destinations to PSTN if (uri=~"^sip:1013@.*$" || uri=~"^sip:[0-9]{10}@.*$" || uri=~"^sip:06[0-9]{8}@.*$") { route(5); route(1); break; }; # native SIP destinations are handled using our USRLOC DB lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); break; } _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

1 0

[Serusers] Re: Welcome to the "Serusers" mailing list (Digest mode)
by gsalvucci＠ing.unrc.edu.ar 07 Jun '05

07 Jun '05

Mensaje citado por serusers-request(a)lists.iptel.org: > Welcome to the serusers(a)lists.iptel.org mailing list! > > To post to this list, send your email to: > > serusers(a)lists.iptel.org > > General information about the mailing list is at: > > http://lists.iptel.org/mailman/listinfo/serusers > > If you ever want to unsubscribe or change your options (eg, switch to > or from digest mode, change your password, etc.), visit your > subscription page at: > > http://lists.iptel.org/mailman/options/serusers/gsalvucci%40ing.unrc.edu.ar > > > You can also make such adjustments via email by sending a message to: > > serusers-request(a)lists.iptel.org > > with the word `help' in the subject or body (don't include the > quotes), and you will get back a message with instructions. > > You must know your password to change your options (including changing > the password, itself) or to unsubscribe. It is: > > hipolita > > Normally, Mailman will remind you of your iptel.org mailing list > passwords once every month, although you can disable this if you > prefer. This reminder will also include instructions on how to > unsubscribe or change your account options. There is also a button on > your options page that will email your current password to you. > > ---------------------------------------------------------------- Este mensaje ha sido enviado usando el servicio de Webmail de la Universidad Nacional de Río Cuarto. Basado en Horde/IMP

1 0

[Serusers] setting up SER
by Tomasz Gingold 07 Jun '05

07 Jun '05

Hello, i´m fairly new in SER. After setting up the SER i tried to register at SER with the kphone. Inside the kphone i set up the Outbound proxy with the adress of the SER. When i try to REGISTER with the kphone at SER i get always also a REGISTER message back from the SER. Kphone SER -----REGISTER---------> <----REGISTER---------- (the first REGISTER follow 3xREGISTER back) <----REGISTER---------- <----REGISTER---------- -----REGISTER---------> <----REGISTER---------- -----REGISTER---------> <----REGISTER---------- -----REGISTER---------> <----REGISTER---------- ... <----403 Timeout------- It looks like the SER doesn't go into the routing logic. I modify the ser.cfg in this way to check this. route { sl_send_reply("420","Bad Extension"); After this nothing has changed in the packet flow described before. No 420 replys are beeing send back. I know that this is not a detailed description of my problem.Anyhow i hope that anybody can give me some tips. thanks in advance tomasz g. -- Geschenkt: 3 Monate GMX ProMail gratis + 3 Ausgaben stern gratis ++ Jetzt anmelden & testen ++ http://www.gmx.net/de/go/promail ++

3 3

[Serusers] Testing with SIP and ENUM
by Adrian Georgescu 07 Jun '05

07 Jun '05

For those of you who need accounts to test their own SER proxies from outside their domains I have setup this easy registration service http://sip2sip.info You get a SIP account under sip2sip plus a UPT +87810 ENUM number and you may call outside local domain with it. There are some mediaproxies that do the NAT traversal, video streams included. And BTW check the SNOM software phone from http://snom.com/download/snom360-3.60j.exe it is by far the easiest SIP software I had configured on my desktop. Regards, Adrian

1 0

[Serusers] help with: can't read reply from a RTP proxy
by jeff kwong 07 Jun '05

07 Jun '05

Hi everybody! Im a newbie with ser and I have just joined the list. You see I want to run a ser+nathelper+rtpproxy system in my linux box. My primary task is to build a sip outbound proxy. As a start I used the nathelper.cfg file in sip_router from the cvs repository. I also downloaded and installed the latest ser and rtpproxy from cvs. I always run rtpproxy first and I can verify its is running with ps. However when I execute ser with nathelper.cfg I keep getting the following message: 0(0) INFO: udp_init: SO_RCVBUF is initially 65535 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 0(0) WARNING: using only the first listen address (no fork) 2(0) INFO: fifo process starting: 15323 1(15322) ERROR: send_rtpp_command: can't read reply from a RTP proxy 1(15322) send_rtpp_command(): proxy </var/run/rtpproxy.sock> does not responding, disable it 1(15322) WARNING: rtpp_test: can't get version of the RTP proxy 1(15322) WARNING: rtpp_test: support for RTP proxy </var/run/rtpproxy.sock>has been disabled temporarily 2(15323) ERROR: send_rtpp_command: can't read reply from a RTP proxy 2(15323) send_rtpp_command(): proxy </var/run/rtpproxy.sock> does not responding, disable it 2(15323) WARNING: rtpp_test: can't get version of the RTP proxy 2(15323) WARNING: rtpp_test: support for RTP proxy </var/run/rtpproxy.sock>has been disabled temporarily 2(15323) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... 2(15323) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(15321) ERROR: send_rtpp_command: can't read reply from a RTP proxy 0(15321) send_rtpp_command(): proxy </var/run/rtpproxy.sock> does not responding, disable it 0(15321) WARNING: rtpp_test: can't get version of the RTP proxy 0(15321) WARNING: rtpp_test: support for RTP proxy </var/run/rtpproxy.sock>has been disabled temporarily and thus I cannot make rtp calls. I know similar problems have been posted before and the most frequent suggestion was to get the latest versions of ser and rtpproxy like the one below. 1) Get ser from CVS, build, install cvs -d:pserver:anonymous at cvs.berlios.de <http://lists.iptel.org/mailman/listinfo/serusers>:/cvsroot/ser co sip_router 2) Get rtpproxy from CVS, build, install cvs -d:pserver:anonymous at cvs.berlios.de <http://lists.iptel.org/mailman/listinfo/serusers>:/cvsroot/ser co rtpproxy 3) Edit ser.cfg to add modparam("nathelper","rtpproxy_sock", "/var/run/rtpproxy.sock") I did as instucted but still the same results. What could I be missing? To those who have successfully configured a similar setup, can I ask what version of ser and rtpproxy are compatible? Thanks in advance! jeff

2 1

[Serusers] Configuration example of lcr.so
by Felipe Martins 07 Jun '05

07 Jun '05

Hi everyone, Has anybody an example of ser.cfg using lcr.so module doing routing logics querying a MySQL DB ? If so, can anyone send it to me ? I have read all the documentation but I'm still confusing about it... Thanks in Advance Best Regards -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

1 0

[Serusers] 4/8 port GW
by Iqbal 07 Jun '05

07 Jun '05

Hi Does anyone have any good experience with 4/8 port gateways and connecting them to SER. If so which are recommended, they must use user/pass to auth I guess. Iqbal

3 2

[Serusers] domain/realm routing
by raid＠gmx.net 07 Jun '05

07 Jun '05

hi all example: uac1 with number 12345 and realm my1.mydomain.com uac2 with number 54321 and realm my2.mydomain.com and so on i use enum to write the called brunch to the realm this funktion is ok (if someone call 12345/123, enum rewrite the uri to 123(a)my1.mydomain.com) but ser says no user 123(a)my1.mydomain.com if i change in enum zonefile my1.mydomain.com to the ip address of the uac ser send the call to the uac1, but there is no connect because the uac1 is behind nat the next problem is if the user has a dynamic ip is there a way to bind an uac/user to an realm (all calls 12345/branches route to uac1(a)my1.mydomain.com) i hope you understand my problem sorry for my bad english :( regards raid -- Weitersagen: GMX DSL-Flatrates mit Tempo-Garantie! Ab 4,99 Euro/Monat: http://www.gmx.net/de/go/dsl

3 4

[Serusers] Loose routing question
by Michael Ulitskiy 07 Jun '05

07 Jun '05

Hello, I'm trying to comprehend loose routing concept and I have a question that concerns me. As far as I understand loose routing says that if there're Route headers in a message it should be forwarded according to the URIs set in Route headers. I thought that this is true only within a dialog, but RFC3261 (part 16.6) says: "Requests establishing a dialog may contain a preloaded Route header field." Also SER manual says: " the failure not to include loose routing in your scripts may lead to infinite loops. Make sure that you include the following script fragment immediately after request sanity checks" and provide the following piece of code: if (loose_route()) { t_relay(); break; }; which as far as I understand unconditionally forwards message if Route header is present. So I'm wondering what about security? If I follow this guidelines how I would shield my PSTN gateway if anyone can construct message and pre-load it with URI of my gateway and all my proxies must honor it. For example I have a PSTN gateway on ip address 10.1.1.5 and proxy on 10.1.1.10 that supposed to interface outside world. So I guess if someone construct a message like this: INVITE sip:12345@somewhere.com SIP/2.0 ... Route: <sip:12345@10.1.1.5;lr> my proxy will forward it to PSTN gateway and it will make outbound call. Is this true? Please enlighten me on this. Thank you, Michael

7 9

[Serusers] do I need to upgrade to SER 0.9?
by Llanos Serna García-Conde 07 Jun '05

07 Jun '05

1 0

[Serusers] MAX_LINE_SIZE
by Llanos Serna García-Conde 07 Jun '05

07 Jun '05

1 0

Re: [Serusers] Can I use realm name without DNS SRV
by Greger V. Teigre 07 Jun '05

07 Jun '05

Sorry, Antonio. This is getting too much of a special case that I can devote any time to it. I would suggest you clean up and concentrate on a pure IP setup (also in the user clients), which, BTW, I have done at several occations, even without an Internet connection. Remember that SER does a reverse lookup (gethostbyname) to resolve it's own IP address. g-) Antonio Cano damas wrote: > Dear Greger, > > I went to fast into this conclusion :( > > I've made rewritehostport("Public_IP:5070") because if I use the > private_ip SER does nothing :? > > The problem now is in Asterisk or Maybe in the network config, I think > is second. > > In my network scenario the Redstar servir is connected to the internet > throught an ADSL Router, with all ports redirected to it: > Internet ----> Public_IP (Router ADSL) Private_IP <---- (Redstar) > > With Tethereal We can see this log: > 11.055288 192.168.1.5 -> Public_IP SIP/SDP Request: INVITE > sip:213@Public_IP:5070, with session description > 11.055982 192.168.1.1 -> 192.168.1.5 ICMP Destination > unreachable (Port unreachable) > > The problem is that packet try to go throw the Internet and the Router > doesn't know how to give it. Due to the Proxy and Asterisk are both > into the same machine I doesn't need that travel :) > > In /etc/hosts I've got: > 192.168.1.5 Public_IP redstar.organization.org redstar > 192.168.1.5redstar.organization.org > > For iptables I've added: > iptables -t nat -A PREROUTING -d Public_IP -p udp --dport 5060:5070 > -j DNAT --to 192.168.1.5 > > After the changes of /etc/hosts I rebooted the server. Now I make some > tests with traceroute: > > redstar:~# traceroute redstar > traceroute to Public_IP (192.168.1.5), 30 hops max, 38 byte packets > 1 80 (192.168.1.5) 0.268 ms 0.058 ms 0.750 ms > > redstar:~# traceroute Public_IP > traceroute to Public_IP (Public_IP), 30 hops max, 38 byte packets > 1 192.168.1.1 (192.168.1.1) 0.634 ms 0.451 ms 0.361 ms > > > Do you have some idea about that? > > Kind regards, > Antonio F. Cano > > > > Greger V. Teigre wrote: > >> ;-) >> Antonio Cano damas wrote: >> >>> wowwwww, a lot of thanks it goes :D >>> >>> >>> Greger V. Teigre wrote: >>> >>>> SER will try to resolve using DNS SRV and A lookups unless you >>>> specify IP addresses, so: >>>> route[2] >>>> { >>>> #Host and port where Asterisk is listening, sip.conf >>>> rewritehostport("192.168.1.5:5070"); >>>> route(1); >>>> break; >>>> } >>>> g-) >>>> >>>> Antonio Cano damas wrote: >>>> >>>>> Hi, >>>>> >>>>> That seems to start working :). I've made the next changes: >>>>> listen="redstar.organization.org" >>>>> alias="192.168.1.5" >>>>> alias="organization.org" >>>>> alias="redstar.organization.org" >>>>> >>>>> check_via=no # (cmd. line: -v) >>>>> dns=no # (cmd. line: -r) >>>>> rev_dns=no # (cmd. line: -R) >>>>> >>>>> And the proper updates into MySQL Database for domain like >>>>> 'redstar.organization.org'. >>>>> >>>>> Now I've got happening something strange, my UAs x-lite and >>>>> grandstream can register but Asterisk doesn't (at first look). >>>>> Looking the URIS I've seen that X-lite generate >>>>> @192.168.1.5redstar.organization.org while Asterisk generate >>>>> @redstar.organization.org. Finally I've got registering the >>>>> Asterisk using fromdomain= 192.168.1.5restar.organization.org :-/ >>>>> >>>>> My next step is try to make a call from X-Lite to one extension of >>>>> Asterisk. In the SER.cfg: >>>>> A) Into the INVITE process section: >>>>> if (uri =~ "^sip:(2[0-9][0-9])@*" ) { >>>>> log(1,"Al asterisk\n"); >>>>> route(2); >>>>> break; >>>>> }; >>>>> B) >>>>> route[2] >>>>> { >>>>> #Host and port where Asterisk is listening, sip.conf >>>>> rewritehostport("192.168.1.5redstar.organization.org:5070"); >>>>> route(1); >>>>> break; >>>>> } >>>>> >>>>> route[1] >>>>> { >>>>> # if client or server know to be behind a NAT, enable relay >>>>> if (isflagset(6)) { >>>>> #log(1, "Pasando por force_rtp_proxy\n"); >>>>> append_hf("P-Hint: Pasando por RTP_PROXY\r\n"); >>>>> force_rtp_proxy(); >>>>> }; >>>>> >>>>> # labeled all transaction for accounting >>>>> #setflag(4); >>>>> >>>>> # send it out now; use stateful forwarding as it works >>>>> reliably # even for UDP2TCP >>>>> if (!t_relay()) { >>>>> sl_reply_error(); >>>>> }; >>>>> } >>>>> >>>>> With that the INVITE petition gets an Request Timeout response, >>>>> making a little of ngrep: >>>>> >>>>> U IP_Client_UA:5060 -> 192.168.1.5:5060 >>>>> INVITE sip:213@192.168.1.5redstar.organization.org SIP/2.0..Via: >>>>> SIP/2.0/UDP >>>>> 192.168.0.6:5060;rport;branch=z9hG4bK4F9444CAD6AD11D9B421000A95A55E26..From: >>>>> >>>>> >>>>> Anto >>>>> nio F. Cano >>>>> <sip:10101@redstar.organization.org>;tag=1817289323..To: >>>>> <sip:213@redstar.organization.org>..Contact: >>>>> <sip:10101@192.168.0.6:5060>..Call-ID: 4F13C7CB-D >>>>> 6AD-11D9-B421-000A95A55E26@192.168.0.6..CSeq: 3733 >>>>> INVITE..Max-Forwards: 70..Content-Type: >>>>> application/sdp..User-Agent: X-Lite release 1103m..Content-Len >>>>> gth: 263....v=0..o=10101 5457996 5458244 IN IP4 >>>>> 192.168.0.6..s=X-Lite..c=IN IP4 192.168.0.6..t=0 0..m=audio 8000 >>>>> RTP/AVP 3 0 8 98 101..a=rtpmap:0 pcmu/80 00..a=rtpmap:8 >>>>> pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 >>>>> iLBC/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 >>>>> 0-15.. >>>>> # >>>>> >>>>> U 192.168.1.5:5060 -> IP_DNS_SERVER:5070 >>>>> INVITE sip:213@192.168.1.5redstar.organization.org:5070 >>>>> SIP/2.0..Record-Route: >>>>> <sip:213@192.168.1.5;ftag=1817289323;lr=on>..Via: SIP/2.0/UDP >>>>> 192.168.1.5;branc h=z9hG4bK151f.afc77ca4.0..Via: SIP/2.0/UDP >>>>> 192.168.0.6:5060;received=IP_Client_UA;rport=5060;branch=z9hG4bK4F9444CAD6AD11D9B421000A95A55E26..From: >>>>> >>>>> >>>>> Antonio >>>>> F. Cano <sip:10101@redstar.organization.org>;tag=1817289323..To: >>>>> <sip:213@redstar.organization.org>..Contact: >>>>> <sip:10101@IP_Client_UA:5060>..Call-ID: 4F13C7CB-D6AD-1 >>>>> 1D9-B421-000A95A55E26@192.168.0.6..CSeq: 3733 >>>>> INVITE..Max-Forwards: 69..Content-Type: application/sdp..User-Agent: >>>>> X-Lite release >>>>> 1103m..Content-Length: >>>>> 283..P-Hint: Nat uac_test=3 ..P-Hint: Metodo INVITE corrigiendo >>>>> SDP..P-Hint: Pasando por RTP_PROXY....v=0..o=10101 5457996 5458244 >>>>> IN IP4 192.168.0.6..s= >>>>> X-Lite..c=IN IP4 192.168.0.6..t=0 0..m=audio 8000 RTP/AVP 3 0 8 >>>>> 98 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 >>>>> gsm/8000..a=rtpmap:98 iLBC >>>>> /8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 >>>>> 0-15..a=direction:active.. >>>>> >>>>> Why SER sends the message to IP_DNS_SERVER in place of >>>>> 192.168.1.5redstar.organization.org? >>>>> >>>>> What's happen here? I don't understand nothing. What I make wrong? >>>>> The @192.168.1.5redstar.organization.org doesn't seem to be so >>>>> elegant is there any way to make it like redstar.organization.org >>>>> >>>>> Kind regards and thanks for your time, >>>>> Antonio F. Cano >>>>> >>>>> Greger V. Teigre wrote: >>>>> >>>>>> You can turn off SRV in X-Lite and Grandstream (Use DNS SRV >>>>>> option) And use SIP proxy: fully qualified domain name (same as >>>>>> outbound) g-) >>>>>> Antonio Cano damas wrote: >>>>>> >>>>>>> Hello all, >>>>>>> >>>>>>> First of all excuses for my bad english. I was looking into the >>>>>>> list and doesn't find nothing similar to my problem maybe I >>>>>>> didn't use the correct keywords. >>>>>>> >>>>>>> I've got a Proxy that is behaind and Public IP and I've got a >>>>>>> subdomain name pointing to this machine. But the DNS server >>>>>>> isn't mine and I can't add a SRV record, I've redirected all >>>>>>> petitions to this server and in that way if I make a petition >>>>>>> of anykind service (ssh, web, ...) to the subdomain.name.com it >>>>>>> comes into the machine. The problem is that all de UA i try >>>>>>> (X-Lite, Grandstream and Asterisk) makes DNS SRV queries and i >>>>>>> don't know how to avoid this. For example: >>>>>>> >>>>>>> My subdomain is redstar.organization.org and I'd like to use the >>>>>>> realm organization.org >>>>>>> >>>>>>> + File /etc/hosts >>>>>>> 192.168.1.5 Public_IP_Of_RedStar redstar.organization.org >>>>>>> redstar >>>>>>> >>>>>>> + In /etc/init.d/ser I've got this params line: >>>>>>> PARAMS="-P $PIDFILE -u root -g root -l >>>>>>> redstar.organization.org" --->I try using -l >>>>>>> organization.org, but the server said to me: >>>>>>> 0(0) ERROR: udp_init: bind(5, 0x80c8cc0, 16) on >>>>>>> IP_Organization.org: Cannot assign requested address >>>>>>> >>>>>>> + In /etc/ser/ser.cfg added this: >>>>>>> alias="organization.org" >>>>>>> alias="redstar.organization.org" >>>>>>> >>>>>>> + MySQL tables the domain field value is 'organization.org' >>>>>>> >>>>>>> When I start the SER it tells me: >>>>>>> Restarting ser: serListening on >>>>>>> Public_IP_Of_Redstar [192.168.1.5]:5060 >>>>>>> Aliases: redstar:5060 redstar.organization.org:* >>>>>>> organization.org:* The UA Grandstream config is: >>>>>>> SIP Server: organization.org >>>>>>> Outbound Proxy: redstar.organization.org >>>>>>> >>>>>>> And the received answer is a 478 Unresolveable destination. Can >>>>>>> anyone help me? What can I do? I'm a little desesperated :( >>>>>>> >>>>>>> Thanks in advance, >>>>>>> Antonio F. Cano >>>>>>> >>>>>>> Ngrep log result into redstar.organization.org machine: >>>>>>> >>>>>>> U UA_Public_IP:62303 -> 192.168.1.5:5060 >>>>>>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: >>>>>>> SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083fbe >>>>>>> 2c60e48e32..From: "Antonio F. Cano (iGT)" >>>>>>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T >>>>>>> o: <sip:10101@organization.org;user=phone>..Contact: >>>>>>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 >>>>>>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: >>>>>>> 3600..User-Agent: Grandstream HT487 1.0.5. >>>>>>> 18..Max-Forwards: 70..Allow: >>>>>>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng >>>>>>> >>>>>>> >>>>>>> th: >>>>>>> 0.... >>>>>>> >>>>>>> # >>>>>>> >>>>>>> Why this 'sip:192.168.1.5organization.org'? >>>>>>> >>>>>>> U 192.168.1.5:32863 -> 195.235.113.3:53 >>>>>>> >>>>>>> Z............_sip._udp.192.168.1.5organization.org..!.. >>>>>>> >>>>>>> >>>>>>> ###################################################################### >>>>>>> >>>>>>> U UA_Public_IP:62303 -> 192.168.1.5:5060 >>>>>>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: >>>>>>> SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083fbe >>>>>>> 2c60e48e32..From: "Antonio F. Cano (iGT)" >>>>>>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T >>>>>>> o: <sip:10101@organization.org;user=phone>..Contact: >>>>>>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 >>>>>>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: >>>>>>> 3600..User-Agent: Grandstream HT487 1.0.5. >>>>>>> 18..Max-Forwards: 70..Allow: >>>>>>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng >>>>>>> >>>>>>> >>>>>>> th: >>>>>>> 0.... >>>>>>> >>>>>>> ######################################################### >>>>>>> U 195.235.113.3:53 -> 192.168.1.5:32863 >>>>>>> >>>>>>> Z............_sip._udp.192.168.1.5organization.org..!...)......*0.=.a.gtld-servers.).nstld.verisign-grs.c >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> om.B..@..........:..... >>>>>>> >>>>>>> # >>>>>>> U 192.168.1.5:32863 -> 195.235.113.3:53 >>>>>>> >>>>>>> Z............_sip._udp.192.168.1.5organization.org..!.. >>>>>>> >>>>>>> ########## >>>>>>> U 195.235.113.3:53 -> 192.168.1.5:32863 >>>>>>> >>>>>>> Z............_sip._udp.192.168.1.5organization.org..!...)......*0.=.a.gtld-servers.).nstld.verisign-grs.c >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> om.B..@..........:..... >>>>>>> >>>>>>> ################################################################################################################## >>>>>>> >>>>>>> >>>>>>> >>>>>>> U UA_Public_IP:62303 -> 192.168.1.5:5060 >>>>>>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: >>>>>>> SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083fbe >>>>>>> 2c60e48e32..From: "Antonio F. Cano (iGT)" >>>>>>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T >>>>>>> o: <sip:10101@organization.org;user=phone>..Contact: >>>>>>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 >>>>>>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: >>>>>>> 3600..User-Agent: Grandstream HT487 1.0.5. >>>>>>> 18..Max-Forwards: 70..Allow: >>>>>>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng >>>>>>> >>>>>>> >>>>>>> th: >>>>>>> 0.... >>>>>>> >>>>>>> ########################################################################################################################### >>>>>>> >>>>>>> >>>>>>> >>>>>>> U 192.168.1.5:5060 -> UA_Public_IP:62303 >>>>>>> SIP/2.0 478 Unresolveable destination (478/TM)..Via: >>>>>>> SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083 >>>>>>> fbe2c60e48e32;rport=62303;received=IP_Client_UA..From: "Antonio >>>>>>> F. Cano (iGT)" <sip:10101@organization.org; >>>>>>> user=phone>;tag=b511698681c5146b..To: >>>>>>> <sip:10101@organization.org;user=phone>;tag=1b908f8c725dfd3e50cbc74 >>>>>>> f548dfb0d-588b..Call-ID: 34ee28e5dad286d9@192.168.0.9..CSeq: >>>>>>> 100 REGISTER..Server: Sip EXpress route >>>>>>> r (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 >>>>>>> 80.38.244.122:5060 "Noisy feedback tells: >>>>>>> pid=8023 req_src_ip=IP_Client_UA req_src_port=62303 >>>>>>> in_uri=sip:192.168.1.5organization.org out_uri=sip:192. >>>>>>> 168.1.5organization.org >>>>>>> via_cnt==1".... >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Serusers mailing list >>>>>>> serusers(a)lists.iptel.org >>>>>>> http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] call forwarding rpid and billing
by Iqbal 07 Jun '05

07 Jun '05

Hi I am trying to make a call from pstn to IP which then forwards out to pstn again It all works fine, except the billing, which should be accounted in the IP account (although logically i cant see how this would work, since the IP user is not strictly speaking making the call) And the second problem is rpid, the rpid is showing the oriniating pstn caller, which is what it should, however if I wanted it to show the IP number instead (i.e the number assigned to that IP account) how could I. Iqbal

3 3

RE: [Serusers] Acc SetFlag issue
by Mr Greg Plater 07 Jun '05

07 Jun '05

Hi I added alias="blah.com" alias="sip.blah.com Tested and same result seen. if the callee hangs up only one BYE is noted in the acc table.. if the caller hangs up I note 3 BYE messages show in the acc table. I have 4 IP phones of different types and no matter what phone starts the call same issue is noted. Thanks for your help _____ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, 7 June 2005 5:10 PM To: Mr Greg Plater Subject: Re: [Serusers] Acc SetFlag issue It could be a loop, but I don't know where all the IPs are coming from. I cannot see alias="domain.com" in your ser.cfg. It's important to the myself check. You must be sure that you have ALL your possible domain names as aliases! g-)

3 5

Re: [Serusers] db_url not working
by Gorka Guridi 07 Jun '05

07 Jun '05

When you compiled SER, are you sure you had uncommented the following line: DEFS+=-DSQL_ACC in Makefile of acc? This file is in ser/modules/acc. If you wish to enable SQL accounting you must do it first.

1 0

Re: [Serusers] Acc SetFlag issue
by Greger V. Teigre 07 Jun '05

07 Jun '05

And you of course t_relay() BYE messages? (I don't think three BYEs in acc table has anything to do with where you put setflag. Either your UACs generate three BYEs because an ACK is not received and you don't t_relay, or something else is seriously wrong...) g-) ---- Original Message ---- From: Mr Greg Plater To: 'Greger V. Teigre' Sent: Tuesday, June 07, 2005 08:23 AM Subject: RE: [Serusers] Acc SetFlag issue > Hi > > I would normally agree with you in this should be simple. > > Tests show same result. when I make a call from one endpoint to > another and the callee hangs up, only a single BYE is noted in the > acc table. When the caller hangs up the call, I get three BYE > messages in the acc table. > > I checked for other setflag(1) and nothing is found. > > Any idea's > > > > > > From: Greger V. Teigre [mailto:greger@teigre.com] > Sent: Tuesday, 7 June 2005 4:00 PM > To: Mr Greg Plater; serusers(a)lists.iptel.org > Subject: Re: [Serusers] Acc SetFlag issue > > This is a fairly straight-forward way to do it: > if (method=="INVITE" || method=="BYE" || method=="ACK") { > setflag(1); > }; > Put it at the top after sanity checks, but before loose routing etc. > g-) > > ---- Original Message ---- > From: Mr Greg Plater > To: serusers(a)lists.iptel.org > Sent: Monday, June 06, 2005 11:41 PM > Subject: [Serusers] Acc SetFlag issue > >> When I have setflag in this location a call logs three times. >> >> Were should I place the setflag if I wish for all calls to log in acc >> with mysql? >> >> >> Ser.cfg >> Snip: >> >> modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") >> modparam("acc", "log_level", 1) >> modparam("acc", "log_flag", 1) >> modparam("acc", "log_fmt", "cdfimorstup") >> modparam("acc", "db_flag", 1) >> modparam("acc", "log_missed_flag", 2) >> modparam("acc", "db_missed_flag", 2) >> modparam("acc", "report_cancels", 1) >> >> >> # ----------------------------------------------------------------- >> # Loose Route Section >> # >> ----------------------------------------------------------------- >> >> #setflag(1); >> >> 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); >> break; >> }; >> >> >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers

2 1

[Serusers] How to download SER
by Cecilia Muaddi 07 Jun '05

07 Jun '05

Hello, I just join the mail list and was hoping someone out there can point me to a location to download SER. It seems to require a user/password from the ftp site ftp://ftp.berlios.de/pub/ser. If so, how do I obtain the user/password. Thanks, Cecilia Muaddi cmuaddi(a)ambernetworks.net Amber Networks 408.626.7300 -------------------------------------------------- The information transmitted is intended only for the person or entity to which it is addressed and may contain confidential and/or privileged material. Any review, retransmission, dissemination or other use of, or taking of any action in reliance upon, this information by persons or entities other than the intended recipient is prohibited. If you received this in error, please contact the sender and delete the material from any computer. This E-mail is covered by the Electronic Communications Privacy Act, 18 U.S.C. §§ 2510-2521 and is legally privileged and/or confidential information. If you are not the intended recipient(s), or the employee or agent responsible for delivery of this message to the intended recipient(s), you are hereby notified that any dissemination, distribution or copying of this e-mail message is strictly prohibited. If you have received this message in error, please immediately notify the sender and delete this e-mail message from your computer.

3 2

[Serusers] ACK not sent
by Greg Martin 07 Jun '05

07 Jun '05

I'm forwarding to a server which returns an invite with changed completion data. I've guarded against loops by assuring the returned invite isn't sent to the server again (I'm not sure that what I'm doing is compliant though). The correct phone rings and when I answer the dialed phone thinks it is connected. The rung phone never does think it's connected though and a packet capture shows no ack sent from the dialed phone which I presume is the problem (the phones are cisco 7960/7940). I'd appreciate any help. Sorry about the long message but felt the ngrep dump and cfg were both necessary. ********************************************************************************** debug=3 fork=yes log_stderror=no dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" 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" modparam("usrloc", "db_mode", 1) modparam("rr", "enable_full_lr", 1) route { # ------------------------------------------------------------------------ # Sanity Check Section # ------------------------------------------------------------------------ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ if (method != "REGISTER") { record_route(); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri != myself) { route(1); break; }; if (method == "REGISTER") { route(2); break; }; if (method == "INVITE") { if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; if(src_ip != "142.178.213.169" && src_ip != "142.178.213.201" && src_ip != "142.178.213.170" && src_ip != "142.178.213.202") { rewritehost("142.178.213.169"); forward(uri:host, uri:port); break; }; } route(1); } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ if (!t_relay()) { sl_reply_error(); }; } route[2] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ if (!save("location")) { sl_reply_error(); }; } *************************************************************************************** interface: qfe0 (142.178.218.0/255.255.254.0) filter: ip and ( port 5060 ) # U 142.178.21.214:50090 -> 142.178.218.210:5060 INVITE sip:5553331@142.178.218.210 SIP/2.0. Via: SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:17 GMT. CSeq: 101 INVITE. User-Agent: CSCO/4. Contact: <sip:5551111@142.178.21.214:5060>. Expires: 180. Content-Type: application/sdp. Content-Length: 249. Accept: application/sdp. . v=0. o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. s=SIP Call. c=IN IP4 142.178.21.214. t=0 0. m=audio 22102 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.213.169:5060 INVITE sip:5553331@142.178.213.169:5060;user=ip SIP/2.0. Max-Forwards: 10. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:17 GMT. CSeq: 101 INVITE. User-Agent: CSCO/4. Contact: <sip:5551111@142.178.21.214:5060>. Expires: 180. Content-Type: application/sdp. Content-Length: 249. Accept: application/sdp. . v=0. o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. s=SIP Call. c=IN IP4 142.178.21.214. t=0 0. m=audio 22102 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. To: <sip:5553331@142.178.218.210>. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 0. . # U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 142.178.21.214:5060. To: <sip:5553331@142.178.218.210>. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 0. . # U 142.178.213.201:37403 -> 142.178.218.210:5060 INVITE sip:5552221@142.178.218.210:5060 SIP/2.0. Via: SIP/2.0/UDP 142.178.213.169:5060;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Max-Forwards: 9. Record-Route: <sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 249. Date: Mon, 06 Jun 2005 22:24:17 GMT. User-Agent: CSCO/4. Contact: <sip:5551111@142.178.21.214:5060>. Expires: 180. Content-Type: application/sdp. Accept: application/sdp. . v=0. o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. s=SIP Call. c=IN IP4 142.178.21.214. t=0 0. m=audio 22102 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 142.178.213.169:5060;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531;received=142.178.213.201. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. To: <sip:5553331@142.178.218.210>. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Server: Sip EXpress router (0.9.1 (sparc64/solaris)). Content-Length: 0. Warning: 392 142.178.218.210:5060 "Noisy feedback tells: pid=14006 req_src_ip=142.178.213.201 req_src_port=37403 in_uri=sip:5552221@142.178.218.210:5060 out_uri=sip:5552221@142.178.21.182:5060;user=ip via_cnt==3". . # U 142.178.218.210:5060 -> 142.178.21.182:5060 INVITE sip:5552221@142.178.21.182:5060;user=ip SIP/2.0. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0. Via: SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Max-Forwards: 8. Record-Route: <sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 249. Date: Mon, 06 Jun 2005 22:24:17 GMT. User-Agent: CSCO/4. Contact: <sip:5551111@142.178.21.214:5060>. Expires: 180. Content-Type: application/sdp. Accept: application/sdp. . v=0. o=Cisco-SIPUA 23215 153 IN IP4 142.178.21.214. s=SIP Call. c=IN IP4 142.178.21.214. t=0 0. m=audio 22102 RTP/AVP 0 8 18 101. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.21.182:52535 -> 142.178.218.210:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:23 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Length: 0. . # U 142.178.21.182:52536 -> 142.178.218.210:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:23 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Length: 0. . # U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:23 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Length: 0. . # U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 0. Date: Mon, 06 Jun 2005 22:24:23 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. . # U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 0. Date: Mon, 06 Jun 2005 22:24:23 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. . # U 142.178.21.182:52537 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.21.182:52538 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.21.182:52539 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.21.182:52540 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.21.182:52541 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=z9hG4bK7fb5.3a62.0,SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.213.201:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.213.169:5060;received=142.178.213.201;branch=z9hG4bKczK75.kDwlvStR8IbIbYqg~~531,SIP/2.0/UDP 142.178.218.210;branch=0,SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:26 GMT. CSeq: 101 INVITE. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. Content-Type: application/sdp. Content-Length: 201. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.213.201:37403 -> 142.178.218.210:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.218.210;branch=0. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.218.210:5060 -> 142.178.21.214:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 142.178.21.214:5060. Record-Route: <sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>,<sip:tc87sv01;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. CSeq: 101 INVITE. Content-Length: 201. Date: Mon, 06 Jun 2005 22:24:26 GMT. Server: CSCO/4. Contact: <sip:5552221@142.178.21.182:5060>. Content-Type: application/sdp. . v=0. o=Cisco-SIPUA 28218 12539 IN IP4 142.178.21.182. s=SIP Call. c=IN IP4 142.178.21.182. t=0 0. m=audio 22646 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 142.178.21.214:50096 -> 142.178.218.210:5060 BYE sip:142.178.218.210:5060;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on SIP/2.0. Via: SIP/2.0/UDP 142.178.21.214:5060. From: "Greg Martin" <sip:5551111@142.178.218.210>;tag=000ded24d863002f5e5cdd96-559e8015. To: <sip:5553331@142.178.218.210>;tag=000ded6c3372000c543313c3-3061c448. Call-ID: 000ded24-d8631dc2-771e744e-2028c718(a)142.178.21.214. Date: Mon, 06 Jun 2005 22:24:31 GMT. CSeq: 102 BYE. User-Agent: CSCO/4. Content-Length: 0. Route: <sip:tc87sv01:5060;lr;osp-pid=d52olkrgzq6w>,<sip:142.178.218.210:5060;ftag=000ded24d863002f5e5cdd96-559e8015;lr=on>, <sip:5552221@142.178.21.182:5060>. . # Greg Martin Technology & Operations TELUS Communication Inc. office: 780-493-2786 cell: 780-718-4139

2 1

Re: [Serusers] Acc SetFlag issue
by Mr Greg Plater 07 Jun '05

07 Jun '05

Hi I would normally agree with you in this should be simple. Tests show same result. when I make a call from one endpoint to another and the callee hangs up, only a single BYE is noted in the acc table. When the caller hangs up the call, I get three BYE messages in the acc table. I checked for other setflag(1) and nothing is found. Any idea's _____ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Tuesday, 7 June 2005 4:00 PM To: Mr Greg Plater; serusers(a)lists.iptel.org Subject: Re: [Serusers] Acc SetFlag issue This is a fairly straight-forward way to do it: if (method=="INVITE" || method=="BYE" || method=="ACK") { setflag(1); }; Put it at the top after sanity checks, but before loose routing etc. g-) ---- Original Message ---- From: Mr Greg Plater To: serusers(a)lists.iptel.org Sent: Monday, June 06, 2005 11:41 PM Subject: [Serusers] Acc SetFlag issue > When I have setflag in this location a call logs three times. > > Were should I place the setflag if I wish for all calls to log in acc > with mysql? > > > Ser.cfg > Snip: > > modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("acc", "log_level", 1) > modparam("acc", "log_flag", 1) > modparam("acc", "log_fmt", "cdfimorstup") > modparam("acc", "db_flag", 1) > modparam("acc", "log_missed_flag", 2) > modparam("acc", "db_missed_flag", 2) > modparam("acc", "report_cancels", 1) > > > # ----------------------------------------------------------------- > # Loose Route Section > # > ----------------------------------------------------------------- > > #setflag(1); > > 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); > break; > }; > > > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Registrering trouble. Perhaps Fifo
by Martin Kjeldsen 07 Jun '05

07 Jun '05

Hi there I'm having trouble registrering with my SER build (0.9.2), and using the auth_mysql ser.cfg from ONsip.org. It is as if SER isn't recieving any of the messages from my UA, and I've been wondering where I might have gone wrong. One thing I've found different from a former build (0.8.14), I have up and running is the socks which SER is listening on. When i do a serctl ps on the 0.9.2 i get this: ser@sipsite:~/running/ser0.9.0/sbin> serctl ps 200 ok 0 22531 attendant 1 22532 fifo server 2 22533 receiver child=0 sock= 212.97.222.xxx:5060 3 22534 receiver child=1 sock= 212.97.222.xxx:5060 4 22535 receiver child=2 sock= 212.97.222.xxx:5060 5 22536 receiver child=3 sock= 212.97.222.xxx:5060 6 22537 timer 7 22538 tcp receiver 8 22539 tcp receiver 9 22540 tcp receiver 10 22541 tcp receiver 11 22542 tcp main process And as far as I can see the same command on 0.8.14 yields almost the same output, except that the sock has a number like '0' and a @ sign is between scok and ip (eg: sock=0 @ ip-adress)! Can this be the cause of my troubles? I'm afraid that I'm not that much into Unix and Sockets! Any and all help will be greatly appreciated! Kind regards Martin

4 10

[Serusers] Acc SetFlag issue
by Mr Greg Plater 07 Jun '05

07 Jun '05

When I have setflag in this location a call logs three times. Were should I place the setflag if I wish for all calls to log in acc with mysql? Ser.cfg Snip: modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "log_fmt", "cdfimorstup") modparam("acc", "db_flag", 1) modparam("acc", "log_missed_flag", 2) modparam("acc", "db_missed_flag", 2) modparam("acc", "report_cancels", 1) # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- #setflag(1); 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); break; };

2 1

[Serusers] Routing Logics based on MySQL Query !!!
by Felipe Martins 07 Jun '05

07 Jun '05

Hi everyone, I am planning to do all my routing logics, based on ser querying my MySQL in order to compare the prefix of the SURI and after that find the right route to perfectly route the call. I use SER 0.9.0 with Mysql Auth. Do anybody know how to do that ? This suggestion was given to me by Iqbal. Thanks in Advance Best Regards -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

2 1

[Serusers] GrandStream BT 100 Voice Echo with SER
by Felipe Martins 07 Jun '05

07 Jun '05

Hi guys, I have a problem with my GrandStream Budge Tone 100. It's REGISTERed with my SER, I can make and receive calls with it but all the calls are voice echoed. Them same doesn't happen at the other softphone or hardphone I'm talking to. I mean, softphone calls to BT100, only BT100 remains with echo, the sofphone has excellent voice quality as bt100 but with no echo . Did anybody have the same problem ?? Thanks in Advance Best Regards -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

3 2

Re: [Serusers] Can I use realm name without DNS SRV
by Greger V. Teigre 07 Jun '05

07 Jun '05

SER will try to resolve using DNS SRV and A lookups unless you specify IP addresses, so: route[2] { #Host and port where Asterisk is listening, sip.conf rewritehostport("192.168.1.5:5070"); route(1); break; } g-) Antonio Cano damas wrote: > Hi, > > That seems to start working :). I've made the next changes: > listen="redstar.organization.org" > alias="192.168.1.5" > alias="organization.org" > alias="redstar.organization.org" > > check_via=no # (cmd. line: -v) > dns=no # (cmd. line: -r) > rev_dns=no # (cmd. line: -R) > > And the proper updates into MySQL Database for domain like > 'redstar.organization.org'. > > Now I've got happening something strange, my UAs x-lite and > grandstream can register but Asterisk doesn't (at first look). > Looking the URIS I've seen that X-lite generate > @192.168.1.5redstar.organization.org while Asterisk generate > @redstar.organization.org. Finally I've got registering the Asterisk > using fromdomain= 192.168.1.5restar.organization.org :-/ > > My next step is try to make a call from X-Lite to one extension of > Asterisk. In the SER.cfg: > A) Into the INVITE process section: > if (uri =~ "^sip:(2[0-9][0-9])@*" ) { > log(1,"Al asterisk\n"); > route(2); > break; > }; > B) > route[2] > { > #Host and port where Asterisk is listening, sip.conf > rewritehostport("192.168.1.5redstar.organization.org:5070"); > route(1); > break; > } > > route[1] > { > # if client or server know to be behind a NAT, enable relay > if (isflagset(6)) { > #log(1, "Pasando por force_rtp_proxy\n"); > append_hf("P-Hint: Pasando por RTP_PROXY\r\n"); > force_rtp_proxy(); > }; > > # labeled all transaction for accounting > #setflag(4); > > # send it out now; use stateful forwarding as it works reliably > # even for UDP2TCP > if (!t_relay()) { > sl_reply_error(); > }; > } > > With that the INVITE petition gets an Request Timeout response, > making a little of ngrep: > > U IP_Client_UA:5060 -> 192.168.1.5:5060 > INVITE sip:213@192.168.1.5redstar.organization.org SIP/2.0..Via: > SIP/2.0/UDP > 192.168.0.6:5060;rport;branch=z9hG4bK4F9444CAD6AD11D9B421000A95A55E26..From: > Anto > nio F. Cano <sip:10101@redstar.organization.org>;tag=1817289323..To: > <sip:213@redstar.organization.org>..Contact: > <sip:10101@192.168.0.6:5060>..Call-ID: 4F13C7CB-D > 6AD-11D9-B421-000A95A55E26@192.168.0.6..CSeq: 3733 > INVITE..Max-Forwards: 70..Content-Type: application/sdp..User-Agent: > X-Lite release 1103m..Content-Len > gth: 263....v=0..o=10101 5457996 5458244 IN IP4 > 192.168.0.6..s=X-Lite..c=IN IP4 192.168.0.6..t=0 0..m=audio 8000 > RTP/AVP 3 0 8 98 101..a=rtpmap:0 pcmu/80 > 00..a=rtpmap:8 pcma/8000..a=rtpmap:3 gsm/8000..a=rtpmap:98 > iLBC/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 > 0-15.. > # > > U 192.168.1.5:5060 -> IP_DNS_SERVER:5070 > INVITE sip:213@192.168.1.5redstar.organization.org:5070 > SIP/2.0..Record-Route: > <sip:213@192.168.1.5;ftag=1817289323;lr=on>..Via: SIP/2.0/UDP > 192.168.1.5;branc h=z9hG4bK151f.afc77ca4.0..Via: SIP/2.0/UDP > 192.168.0.6:5060;received=IP_Client_UA;rport=5060;branch=z9hG4bK4F9444CAD6AD11D9B421000A95A55E26..From: > Antonio > F. Cano <sip:10101@redstar.organization.org>;tag=1817289323..To: > <sip:213@redstar.organization.org>..Contact: > <sip:10101@IP_Client_UA:5060>..Call-ID: 4F13C7CB-D6AD-1 > 1D9-B421-000A95A55E26@192.168.0.6..CSeq: 3733 INVITE..Max-Forwards: > 69..Content-Type: application/sdp..User-Agent: X-Lite release > 1103m..Content-Length: > 283..P-Hint: Nat uac_test=3 ..P-Hint: Metodo INVITE corrigiendo > SDP..P-Hint: Pasando por RTP_PROXY....v=0..o=10101 5457996 5458244 IN > IP4 192.168.0.6..s= > X-Lite..c=IN IP4 192.168.0.6..t=0 0..m=audio 8000 RTP/AVP 3 0 8 98 > 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rtpmap:3 > gsm/8000..a=rtpmap:98 iLBC > /8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 > 0-15..a=direction:active.. > > Why SER sends the message to IP_DNS_SERVER in place of > 192.168.1.5redstar.organization.org? > > What's happen here? I don't understand nothing. What I make wrong? The > @192.168.1.5redstar.organization.org doesn't seem to be so elegant is > there any way to make it like redstar.organization.org > > Kind regards and thanks for your time, > Antonio F. Cano > > Greger V. Teigre wrote: > >> You can turn off SRV in X-Lite and Grandstream (Use DNS SRV option) >> And use SIP proxy: fully qualified domain name (same as outbound) >> g-) >> Antonio Cano damas wrote: >> >>> Hello all, >>> >>> First of all excuses for my bad english. I was looking into the list >>> and doesn't find nothing similar to my problem maybe I didn't use >>> the correct keywords. >>> >>> I've got a Proxy that is behaind and Public IP and I've got a >>> subdomain name pointing to this machine. But the DNS server isn't >>> mine and I can't add a SRV record, I've redirected all petitions to >>> this server and in that way if I make a petition of anykind service >>> (ssh, web, ...) to the subdomain.name.com it comes into the >>> machine. The problem is that all de UA i try (X-Lite, Grandstream and >>> Asterisk) makes DNS SRV queries and i don't know how to avoid this. >>> >>> For example: >>> >>> My subdomain is redstar.organization.org and I'd like to use the >>> realm organization.org >>> >>> + File /etc/hosts >>> 192.168.1.5 Public_IP_Of_RedStar redstar.organization.org >>> redstar >>> >>> + In /etc/init.d/ser I've got this params line: >>> PARAMS="-P $PIDFILE -u root -g root -l >>> redstar.organization.org" --->I try using -l >>> organization.org, but the server said to me: 0(0) >>> ERROR: udp_init: bind(5, 0x80c8cc0, 16) on IP_Organization.org: >>> Cannot assign requested address >>> >>> + In /etc/ser/ser.cfg added this: >>> alias="organization.org" >>> alias="redstar.organization.org" >>> >>> + MySQL tables the domain field value is 'organization.org' >>> >>> When I start the SER it tells me: >>> Restarting ser: serListening on >>> Public_IP_Of_Redstar [192.168.1.5]:5060 >>> Aliases: redstar:5060 redstar.organization.org:* >>> organization.org:* The UA Grandstream config is: >>> SIP Server: organization.org >>> Outbound Proxy: redstar.organization.org >>> >>> And the received answer is a 478 Unresolveable destination. Can >>> anyone help me? What can I do? I'm a little desesperated :( >>> >>> Thanks in advance, >>> Antonio F. Cano >>> >>> Ngrep log result into redstar.organization.org machine: >>> >>> U UA_Public_IP:62303 -> 192.168.1.5:5060 >>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP >>> 192.168.0.9:62303;branch=z9hG4bK083fbe >>> 2c60e48e32..From: "Antonio F. Cano (iGT)" >>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T >>> o: <sip:10101@organization.org;user=phone>..Contact: >>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 >>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: >>> 3600..User-Agent: Grandstream HT487 1.0.5. >>> 18..Max-Forwards: 70..Allow: >>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng >>> th: >>> 0.... >>> >>> # >>> >>> Why this 'sip:192.168.1.5organization.org'? >>> >>> U 192.168.1.5:32863 -> 195.235.113.3:53 >>> >>> Z............_sip._udp.192.168.1.5organization.org..!.. >>> >>> >>> ###################################################################### >>> U UA_Public_IP:62303 -> 192.168.1.5:5060 >>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP >>> 192.168.0.9:62303;branch=z9hG4bK083fbe >>> 2c60e48e32..From: "Antonio F. Cano (iGT)" >>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T >>> o: <sip:10101@organization.org;user=phone>..Contact: >>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 >>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: >>> 3600..User-Agent: Grandstream HT487 1.0.5. >>> 18..Max-Forwards: 70..Allow: >>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng >>> th: >>> 0.... >>> >>> ######################################################### >>> U 195.235.113.3:53 -> 192.168.1.5:32863 >>> >>> Z............_sip._udp.192.168.1.5organization.org..!...)......*0.=.a.gtld-servers.).nstld.verisign-grs.c >>> >>> >>> om.B..@..........:..... >>> >>> # >>> U 192.168.1.5:32863 -> 195.235.113.3:53 >>> >>> Z............_sip._udp.192.168.1.5organization.org..!.. >>> >>> ########## >>> U 195.235.113.3:53 -> 192.168.1.5:32863 >>> >>> Z............_sip._udp.192.168.1.5organization.org..!...)......*0.=.a.gtld-servers.).nstld.verisign-grs.c >>> >>> >>> om.B..@..........:..... >>> >>> ################################################################################################################## >>> >>> U UA_Public_IP:62303 -> 192.168.1.5:5060 >>> REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP >>> 192.168.0.9:62303;branch=z9hG4bK083fbe >>> 2c60e48e32..From: "Antonio F. Cano (iGT)" >>> <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T >>> o: <sip:10101@organization.org;user=phone>..Contact: >>> <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 >>> 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: >>> 3600..User-Agent: Grandstream HT487 1.0.5. >>> 18..Max-Forwards: 70..Allow: >>> INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng >>> th: >>> 0.... >>> >>> ########################################################################################################################### >>> >>> U 192.168.1.5:5060 -> UA_Public_IP:62303 >>> SIP/2.0 478 Unresolveable destination (478/TM)..Via: SIP/2.0/UDP >>> 192.168.0.9:62303;branch=z9hG4bK083 >>> fbe2c60e48e32;rport=62303;received=IP_Client_UA..From: "Antonio F. >>> Cano (iGT)" <sip:10101@organization.org; >>> user=phone>;tag=b511698681c5146b..To: >>> <sip:10101@organization.org;user=phone>;tag=1b908f8c725dfd3e50cbc74 >>> f548dfb0d-588b..Call-ID: 34ee28e5dad286d9@192.168.0.9..CSeq: 100 >>> REGISTER..Server: Sip EXpress route >>> r (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 >>> 80.38.244.122:5060 "Noisy feedback tells: >>> pid=8023 req_src_ip=IP_Client_UA req_src_port=62303 >>> in_uri=sip:192.168.1.5organization.org out_uri=sip:192. >>> 168.1.5organization.org >>> via_cnt==1".... >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] Sip gateways
by Clay Bryan 07 Jun '05

07 Jun '05

I have recently found out that our current gateway that we use for our pstn will not do both h323 and sip at the same time. I am looking for a way to possible sip trunk from my Cisco Callmanger to ser or just use a cisco gateway to do it. I am not so sure about the cisco gateway supporting both h323 and sip. I would appreciate any advice on this situation. Thanks in advance. Wercs Communications Clay Bryan Network Administrator Wercs Communications 400 East First Casper, WY 82601 <http://maps.yahoo.com/py/maps.py?Pyt=Tmap&addr=400+East+First&csz=Caspe r%2C+WY++82601&country=us> CBryan(a)wercs.com <mailto:CBryan@wercs.com> tel: fax: mobile: 307-233-8359 307-233-8701 307-258-7371 Add me to your address book... <https://www.plaxo.com/add_me?u=21475170292&v0=646557&k0=282754785> Want a signature like this? <http://www.plaxo.com/signature>

1 0

[Serusers] ldap+ser
by Alberto Aguilar 07 Jun '05

07 Jun '05

I heard somebody wanted to use LDAP instead of MySQL, currently Im developing a module to replace MySQL database with an LDAP directory. If you are interested please check the following URL: http://rocksteady.cs.fiu.edu/SER-LDAP/ Regards, Alberto

4 3

[Serusers] SIP_DOMAIN again
by Juan 06 Jun '05

06 Jun '05

Hi all. I just have my SER running on a FreeBSD machine. I put the export SIP_DOMAIN= 10.0.1.10 line in all the .profile and profile files I found but I still have the same "You need to set...."phrase when triyng to run serctl I also tried to do export from the console but I got export:command not found. The shell is csh. Any help will be appreciated. Juan Ferrari

2 2

[Serusers] configuration file
by Llanos Serna García-Conde 06 Jun '05

06 Jun '05

2 1

[Serusers] Sems-voicemail Vs Asterisk-voicemail
by Ahmed Ahmed 06 Jun '05

06 Jun '05

Hello all, I got SER working, to add voicemail service. I need a help which one is better and why Sems-voicemail or Asterisk-voicemail. Thanks Ahmed __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] can't open greeting
by Llanos Serna García-Conde 06 Jun '05

06 Jun '05

1 0

RE: [Serusers] Mediaproxy in LAN (?)
by Giudice, Salvatore 06 Jun '05

06 Jun '05

Mediaproxy will be between the clients for RTP only. Signaling will be relayed by SER. -----Original Message----- From: a cairo [mailto:a.cairo@scsitalia.it] Sent: Saturday, June 04, 2005 6:14 AM To: Giudice, Salvatore; bogdan aux; serusers(a)lists.iptel.org Subject: RE: [Serusers] Mediaproxy in LAN (?) Hi all, i don't understand architecture of mediaproxy with ser ! Is mediaproxy between client and ser? Thanks -----Original Message----- From: "Giudice, Salvatore" <Salvatore.Giudice(a)FMR.COM> To: "bogdan aux" <aux1d(a)yahoo.com>, <serusers(a)lists.iptel.org> Date: Tue, 17 May 2005 20:10:46 -0400 Subject: RE: [Serusers] Mediaproxy in LAN (?) > I am trying to route a call from a non-routable 10.x.x.x network to a > phone on the internet using ser 0.9.0 and mediaproxy 1.31. Aka: PSTN to > SIP from my asterisk to an x-lite > > I found some references indicating that I should call: > > > if ( src_ip = IP_GATEWAY) > { > force_rport(); > fix_contact(); > use_media_proxy(); > }; > > Supposedly, you need the force_rport and the fix_contact to put media > proxy in the middle of the two rtp sessions, but I have et to see this > actually work. Has anyone been able to pass a call through media proxy > from a GW on a non-routable to a routable network, or even between two > ip's on the same network as the mediaproxy? > > If so, please share your configs. > > -----Original Message----- > From: bogdan aux [mailto:aux1d@yahoo.com] > Sent: Friday, May 13, 2005 6:16 AM > To: serusers(a)lists.iptel.org > Subject: [Serusers] Mediaproxy in LAN (?) > > I want to route RTP packets between 2 computers in the > same network through Mediaproxy. I know this may have > no sense, but it's just a test configuration. > The call use_media_proxy() seems to have no effect, > the RTP packets go directly from one computer to > another. > My guess is mediaproxy makes a test on the IPs (they > are 192.168.147.2 and 192.168.147.3) and refuses to > proxy the call. > Is mediaproxy working just between different networks > or it's an error in my configuration file? > > > > > Yahoo! Mail > Stay connected, organized, and protected. Take the tour: > http://tour.mail.yahoo.com/mailtour.html > > _______________________________________________ > 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

1 0

[Serusers] rewrite sip status
by Juan Priotti 06 Jun '05

06 Jun '05

Is it possible to rewrite a sip status received from a gateway? I mean, if I get a 183 Session Progress I want to rewrite it to a 180 Ringing. I've tried using the replace function but it only works for the "From" and "To" fiels. This is what I've used: onreply_route[1] { if (status =~ "183") { log(1, "183 received"); replace("183 Session Progress", "180 Ringing"); } Thank you Juan

1 0

[Serusers] Too Many Hops on Route :(
by Ozan Blotter 06 Jun '05

06 Jun '05

Dear Friends, This is fifth or tenth times writing to list but still no way :( REGISTERED USERS (12345 and 54321 are ATAs, 201 is Analog FXO's Port 0) [root@localhost ser]# serctl ul show Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'location' size : 512 table: 0xf50cb0f8 d_ll { n : 3 first: 0xf50cd108 last : 0xf50cd388 } ...Record(0xf50cd108)... domain: 'location' aor : '12345' ~~~Contact(0xf50cd168)~~~ domain : 'location' aor : '12345' Contact : 'sip:12345@192.168.1.200:5060' Expires : 33 q : Call-ID : '410D1B912D91111(a)192.168.1.200' CSeq : 7962 User-Agent: 'Unknown' received : '' State : CS_NEW Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0xf50cd248)... domain: 'location' aor : '54321' ~~~Contact(0xf50cd2a8)~~~ domain : 'location' aor : '54321' Contact : 'sip:54321@192.168.1.201:5060' Expires : 34 q : Call-ID : '379D1B912DA1111(a)192.168.1.201' CSeq : 7955 User-Agent: 'Unknown' received : '' State : CS_NEW Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ...Record(0xf50cd388)... domain: 'location' aor : '201' ~~~Contact(0xf50cd3e8)~~~ domain : 'location' aor : '201' Contact : 'sip:201@192.168.1.35:5060' Expires : 37 q : Call-ID : 'c0a80123-13c4-3a570cf4-1f6-5973' CSeq : 184 User-Agent: 'Unknown' received : '' State : CS_NEW Flags : 0 next : (nil) prev : (nil) ~~~/Contact~~~~ .../Record... ---/Domain--- ---Domain--- name : 'aliases' size : 512 table: 0xf50c9068 d_ll { n : 0 first: (nil) last : (nil) } ---/Domain--- ===/Domain list=== (Basic SER.CFG , I want to it to route any 053XXXXXXXX Calls to 192.168.1.35 [Analog FXO Gateway] by adding 201 Prefix so it dials a Cellular. Nearly same thing worked for Brekeke's Ondo SIP Proxy but why id does not work in SER) debug=3 fork=yes log_stderror=no listen=192.168.1.10 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" 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" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) route { # ------------------------------------------------------------------------ # Sanity Check Section # ------------------------------------------------------------------------ if (!mf_process_maxfwd_header("12")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ------------------------------------------------------------------------ # Record Route Section # ------------------------------------------------------------------------ if (method!="REGISTER") { record_route(); }; # ------------------------------------------------------------------------ # Loose Route Section # ------------------------------------------------------------------------ if (loose_route()) { route(1); break; }; # ------------------------------------------------------------------------ # Call Type Processing Section # ------------------------------------------------------------------------ if (uri!=myself) { route(1); break; }; if (uri==myself) { if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (uri=~"^sip:053[0-9]*@.*") { prefix("201"); rewritehostport ("192.168.1.35:5060"); # forward ("192.168.1.35:5060"); # t_relay_to_udp("192.168.1.35","5060"); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }; } route[1] { # ------------------------------------------------------------------------ # Default Message Handler # ------------------------------------------------------------------------ if (!t_relay()) { sl_reply_error(); }; } route[2] { # ------------------------------------------------------------------------ # REGISTER Message Handler # ------------------------------------------------------------------------ if (!save("location")) { sl_reply_error(); }; } SIPSAK RESULTS (Too Many Hops and User Not Found Errors) [root@localhost ser]# sipsak -T -s sip:20105353490056@192.168.1.10 warning: IP extract from warning activated to be more informational 0: 192.168.1.10 (0.098 ms) SIP/2.0 483 Too Many Hops 1: 192.168.1.10 (0.244 ms) SIP/2.0 404 User Not Found without Contact header [root@localhost ser]# sipsak -T -s sip:12345@192.168.1.10 warning: IP extract from warning activated to be more informational 0: 192.168.1.10 (0.084 ms) SIP/2.0 483 Too Many Hops 1: ?? (13.000 ms) SIP/2.0 200 OK Contact: <sip:12345@192.168.1.200:5060> [root@localhost ser]# sipsak -T -s sip:54321@192.168.1.10 warning: IP extract from warning activated to be more informational 0: 192.168.1.10 (0.079 ms) SIP/2.0 483 Too Many Hops 1: ?? (13.020 ms) SIP/2.0 200 OK Contact: <sip:54321@192.168.1.201:5060> [root@localhost ser]#

2 1

[Serusers] ruri rewrite question
by Medve Istvan 06 Jun '05

06 Jun '05

Hi All, if I rewrite ruri in the INVITE message I have to rewrite ruri in any subsequent messages (CANCEL, BYE, ACK) in the same dialog ? Or rewrite INVITE is enough ? thx, imedve

2 1

[Serusers] Can I use realm name without DNS SRV
by Antonio Cano damas 06 Jun '05

06 Jun '05

Hello all, First of all excuses for my bad english. I was looking into the list and doesn't find nothing similar to my problem maybe I didn't use the correct keywords. I've got a Proxy that is behaind and Public IP and I've got a subdomain name pointing to this machine. But the DNS server isn't mine and I can't add a SRV record, I've redirected all petitions to this server and in that way if I make a petition of anykind service (ssh, web, ...) to the subdomain.name.com it comes into the machine. The problem is that all de UA i try (X-Lite, Grandstream and Asterisk) makes DNS SRV queries and i don't know how to avoid this. For example: My subdomain is redstar.organization.org and I'd like to use the realm organization.org + File /etc/hosts 192.168.1.5 Public_IP_Of_RedStar redstar.organization.org redstar + In /etc/init.d/ser I've got this params line: PARAMS="-P $PIDFILE -u root -g root -l redstar.organization.org" --->I try using -l organization.org, but the server said to me: 0(0) ERROR: udp_init: bind(5, 0x80c8cc0, 16) on IP_Organization.org: Cannot assign requested address + In /etc/ser/ser.cfg added this: alias="organization.org" alias="redstar.organization.org" + MySQL tables the domain field value is 'organization.org' When I start the SER it tells me: Restarting ser: serListening on Public_IP_Of_Redstar [192.168.1.5]:5060 Aliases: redstar:5060 redstar.organization.org:* organization.org:* The UA Grandstream config is: SIP Server: organization.org Outbound Proxy: redstar.organization.org And the received answer is a 478 Unresolveable destination. Can anyone help me? What can I do? I'm a little desesperated :( Thanks in advance, Antonio F. Cano Ngrep log result into redstar.organization.org machine: U UA_Public_IP:62303 -> 192.168.1.5:5060 REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083fbe 2c60e48e32..From: "Antonio F. Cano (iGT)" <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T o: <sip:10101@organization.org;user=phone>..Contact: <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: 3600..User-Agent: Grandstream HT487 1.0.5. 18..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng th: 0.... # Why this 'sip:192.168.1.5organization.org'? U 192.168.1.5:32863 -> 195.235.113.3:53 Z............_sip._udp.192.168.1.5organization.org..!.. ###################################################################### U UA_Public_IP:62303 -> 192.168.1.5:5060 REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083fbe 2c60e48e32..From: "Antonio F. Cano (iGT)" <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T o: <sip:10101@organization.org;user=phone>..Contact: <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: 3600..User-Agent: Grandstream HT487 1.0.5. 18..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng th: 0.... ######################################################### U 195.235.113.3:53 -> 192.168.1.5:32863 Z............_sip._udp.192.168.1.5organization.org..!...)......*0.=.a.gtld-servers.).nstld.verisign-grs.c om.B..@..........:..... # U 192.168.1.5:32863 -> 195.235.113.3:53 Z............_sip._udp.192.168.1.5organization.org..!.. ########## U 195.235.113.3:53 -> 192.168.1.5:32863 Z............_sip._udp.192.168.1.5organization.org..!...)......*0.=.a.gtld-servers.).nstld.verisign-grs.c om.B..@..........:..... ################################################################################################################## U UA_Public_IP:62303 -> 192.168.1.5:5060 REGISTER sip:192.168.1.5organization.org SIP/2.0..Via: SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083fbe 2c60e48e32..From: "Antonio F. Cano (iGT)" <sip:10101@organization.org;user=phone>;tag=b511698681c5146b..T o: <sip:10101@organization.org;user=phone>..Contact: <sip:10101@192.168.0.9:62303;user=phone>..Call-ID: 3 4ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Expires: 3600..User-Agent: Grandstream HT487 1.0.5. 18..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Leng th: 0.... ########################################################################################################################### U 192.168.1.5:5060 -> UA_Public_IP:62303 SIP/2.0 478 Unresolveable destination (478/TM)..Via: SIP/2.0/UDP 192.168.0.9:62303;branch=z9hG4bK083 fbe2c60e48e32;rport=62303;received=80.32.90.5..From: "Antonio F. Cano (iGT)" <sip:10101@organization.org; user=phone>;tag=b511698681c5146b..To: <sip:10101@organization.org;user=phone>;tag=1b908f8c725dfd3e50cbc74 f548dfb0d-588b..Call-ID: 34ee28e5dad286d9@192.168.0.9..CSeq: 100 REGISTER..Server: Sip EXpress route r (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 80.38.244.122:5060 "Noisy feedback tells: pid=8023 req_src_ip=80.32.90.5 req_src_port=62303 in_uri=sip:192.168.1.5organization.org out_uri=sip:192. 168.1.5organization.org via_cnt==1"....

2 1

[Serusers] Problem with installing SER. Howto SER Step by Step?
by MapsAir 06 Jun '05

06 Jun '05

Dear Helpers! I am very new to linux and SER. I have read the article on your website and am very interested in learning about the sip proxy SER. However, I am still unable to install the SER. Would you please tell me how to install the SER or directly me to an article of step by step how to install the SER. I have read you how to. However, I'm pretty now to linux, and I am still unable to install it. I would be very appreciated for your time! Thanks in advance! Lan Dear Helpers! I am very new to linux and SER. I have read the article on your website and am very interested in learning about the sip proxy SER. However, I am still unable to install the SER. Would you please tell me how to install the SER or directly me to an article of step by step how to install the SER. I have read you how to. However, I'm pretty now to linux, and I am still unable to install it. I would be very appreciated for your time! Thanks in advance! Lan

3 2

[Serusers] rpid again
by Dirk Willbrandt 06 Jun '05

06 Jun '05

Hi list ! I have a Problem with the appending of an rpid with the functions avp_load_radius(), append_rpid_hf(). In my Radius DB i have an SIP-AVP field which is filled with the Value rpid:[NUMBER] When i run manuall the Radius Reply-Query i get the correct value for SIP-AVP. Now, i reconfigured my ser.cfg so that an avp_load_radius() is executed after an incoming INVITE. My Radius is answering on the command and sends the correct SIP-AVP Value for the User. Now my Problem: I can't append the RPID or anything else to the SIP-Header after the answer from the Radius. So the call is going out without the rpid in the Header. Can anyone help me ? Here is the Configure-Snip: if (method=="INVITE") { # here comes the invite if (!radius_proxy_authorize("")) { # for the digest avp_load_radius it seems that i must use an proxy auth proxy_challenge("","0"); # and an proxy challenge if(!avp_load_radius("digest")) { # here i get the SIP-AVP from the radius log(1,"Failure with the AVP-Value\n"); # if it fails i'll get a message break; # go out } avp_print(); # print all avp's (this command is never executed :( ) append_hf ("Remote-Party-ID: 111222333\r\n"); # append this silly row - it is never executed append_rpid_hf("", ";party=calling;id-type=subscriber;screen=yes"); # append the rpid from SIP-AVP - and this command is also never executed break; # go out }; route(3); # go to route block 3 break; # end } I tried also this snippet without the breaks and proxy-commands but it doesn't work. I don't know what is wrong - please help me :s Thanks ! Dirk

3 2

Re: [Serusers] ser don't start nedd help
by harry gaillac 06 Jun '05

06 Jun '05

I drop create ser db. However when I start ser i get : serveur1:~# tail /var/log/syslog Jun 3 18:39:01 serveur1 /USR/SBIN/CRON[2616]: (root) CMD ( [ -d /var/lib/php4 ] && find /var/lib/php4/ -type f -cmin +$(/usr/lib/php4/maxlifetime) -print0 | xargs -r -0 rm) Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: Maxfwd module- initializing Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: permissions - initializing Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: WARNING: File not found: /etc/ser/permissions.allow Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: Default allow file (/etc/ser/permissions.allow) not found => empty rule set Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: WARNING: File not found: /etc/ser/permissions.deny Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: Default deny file (/etc/ser/permissions.deny) not found => empty rule set Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: AVPops - initializing Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: auth_db:str_fixup: Invalid table version (use ser_mysql.sh reinstall) Jun 3 18:44:02 serveur1 /usr/sbin/ser[2729]: ERROR: fix_expr : fix_actions error version table from phpmyadmin: Résultat de la requête SQL Serveur: localhost Base de données: ser Généré le : Vendredi 03 Juin 2005 à 18:39 Généré par: phpMyAdmin 2.6.2 / MySQL 4.0.24_Debian-10-log requête SQL: SELECT * FROM `version` LIMIT 0, 30; Enregistrements: 27 table_name table_version subscriber 5 reserved 1 phonebook 1 pending 4 missed_calls 2 location 6 grp 2 event 1 aliases 6 active_sessions 1 acc 2 config 1 silo 3 realm 1 domain 1 uri 1 server_monitoring 1 server_monitoring_agg 1 trusted 1 usr_preferences 2 usr_preferences_types 1 admin_privileges 1 calls_forwarding 1 speed_dial 2 gw 1 gw_grp 1 lcr 1 Version table with command: +-----------------------+---------------+ | table_name | table_version | +-----------------------+---------------+ | subscriber | 5 | | reserved | 1 | | phonebook | 1 | | pending | 4 | | missed_calls | 2 | | location | 6 | | grp | 2 | | event | 1 | | aliases | 6 | | active_sessions | 1 | | acc | 2 | | config | 1 | | silo | 3 | | realm | 1 | | domain | 1 | | uri | 1 | | server_monitoring | 1 | | server_monitoring_agg | 1 | | trusted | 1 | | usr_preferences | 2 | | usr_preferences_types | 1 | | admin_privileges | 1 | | calls_forwarding | 1 | | speed_dial | 2 | | gw | 1 | | gw_grp | 1 | | lcr | 1 | +-----------------------+---------------+ 27 rows in set (0.01 sec) --- Iqbal <iqbal(a)gigo.co.uk> a écrit : > they should be identical, do u keep logs of queries > sent to Mysql , if > so look at them when u use phpmyadmin, and also > manual queries, and you > will see what is actually being sent, the log will > usually be in > /var/log on the machine with the DB > > Iqbal > > harry gaillac wrote: > > >Why datas from phpmyadmin are not equal to > >mysql> select * from version; > > > >Is it a bug ? > >--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > > > > > > > >>whats the error, I cut and pasted that query and > it > >>deleted the row, > >> > >>Iqbal > >> > >>harry gaillac wrote: > >> > >> > >> > >>>Do you know the right SQL command to delete > >>> > >>> > >>multiple > >> > >> > >>>values ? > >>>I try > >>>mysql> delete from version where > >>> > >>> > >>table_name='trusted'; > >> > >> > >>>it's not good > >>> > >>>Harry > >>> > >>>--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > >>> > >>> > >>> > >>> > >>> > >>>>what does select * from version display > >>>> > >>>>Iqbal > >>>> > >>>>harry gaillac wrote: > >>>> > >>>> > >>>> > >>>> > >>>> > >>>>>Ok, > >>>>> > >>>>>I understand . > >>>>>I use phpmyadmin to manage ser however > >>>>>mysql> select * from version; don't display the > >>>>> > >>>>> > >>>>> > >>>>> > >>>>same > >>>> > >>>> > >>>> > >>>> > >>>>>result > >>>>> > >>>>>Harry > >>>>> > >>>>>--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>>>now remove all the duplicate rows, leaving > only > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>one > >>>> > >>>> > >>>> > >>>> > >>>>>>of each type, eg if > >>>>>>you have alook there are two rows called > >>>>>>domains...remove one, lcr is > >>>>>>also there twice, > >>>>>> > >>>>>>Iqbal > >>>>>> > >>>>>>harry gaillac wrote: > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>>>here is table version from ser DB: > >>>>>>>+-----------------------+---------------+ > >>>>>>>| subscriber | 5 | > >>>>>>>| reserved | 1 | > >>>>>>>| phonebook | 1 | > >>>>>>>| pending | 4 | > >>>>>>>| missed_calls | 2 | > >>>>>>>| location | 6 | > >>>>>>>| grp | 2 | > >>>>>>>| event | 1 | > >>>>>>>| aliases | 6 | > >>>>>>>| active_sessions | 1 | > >>>>>>>| acc | 2 | > >>>>>>>| config | 1 | > >>>>>>>| silo | 3 | > >>>>>>>| realm | 1 | > >>>>>>>| domain | 1 | > >>>>>>>| uri | 1 | > >>>>>>>| server_monitoring | 1 | > >>>>>>>| server_monitoring_agg | 1 | > >>>>>>>| trusted | 1 | > >>>>>>>| usr_preferences | 2 | > >>>>>>>| usr_preferences_types | 1 | > >>>>>>>| admin_privileges | 1 | > >>>>>>>| calls_forwarding | 1 | > >>>>>>>| speed_dial | 2 | > >>>>>>>| gw | 1 | > >>>>>>>| gw_grp | 1 | > >>>>>>>| lcr | 1 | > >>>>>>>| subscriber | 5 | > >>>>>>>| reserved | 1 | > >>>>>>>| phonebook | 1 | > >>>>>>>| pending | 4 | > >>>>>>>| missed_calls | 2 | > >>>>>>>| location | 6 | > >>>>>>>| grp | 2 | > >>>>>>>| event | 1 | > >>>>>>>| aliases | 6 | > >>>>>>>| active_sessions | 1 | > >>>>>>>| acc | 2 | > >>>>>>>| config | 1 | > >>>>>>>| silo | 3 | > >>>>>>>| domain | 1 | > >>>>>>>| domain | 1 | > >>>>>>>| uri | 1 | > >>>>>>>| server_monitoring | 1 | > >>>>>>>| server_monitoring_agg | 1 | > >>>>>>>| trusted | 1 | > >>>>>>>| usr_preferences | 2 | > >>>>>>>| usr_preferences_types | 1 | > >>>>>>>| admin_privileges | 1 | > >>>>>>>| calls_forwarding | 1 | > >>>>>>>| speed_dial | 2 | > >>>>>>>| gw | 1 | > >>>>>>>| gw_grp | 1 | > >>>>>>>| lcr | 1 | > >>>>>>>+-----------------------+---------------+ > >>>>>>>54 rows in set (0.00 sec) > >>>>>>> > >>>>>>>--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>>>can u dump your version table here, and let > us > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>>> > >>>>>>look > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>>>>at it > === message truncated === _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

4 13

[Serusers] SER 0.9.2?
by Dennis Real 06 Jun '05

06 Jun '05

Hi, where can I get SER Version 0.9.2? What switch do I need for cvs? regards

2 1

[Serusers] RE: Re: radiusclient
by Kamran Ahmad 06 Jun '05

06 Jun '05

any comments how to reslove this issue auth_radius.so>: /usr/local/lib/ser/modules/auth_radius.so: undefined symbol: fm_malloc May 26 16:32:04 voice ser: parse error (25,13-14): failed to load module May 26 16:32:04 voice ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/uri_radius.so>: /usr/local/lib/ser/modules/uri_radius.so: undefined symbol: fm_malloc May 26 16:32:04 voice ser: parse error (35,13-14): failed to load module May 26 16:32:04 voice ser: set_mod_param_regex: No module matching auth_radius found | May 26 16:32:04 voice ser: parse error (48,18-19): Can't set module parameter May 26 16:32:04 voice ser: set_mod_param_regex: No module matching auth_radius found | May 26 16:32:04 voice ser: parse error (49,18-19): Can't set module parameter __________________________________ Discover Yahoo! Stay in touch with email, IM, photo sharing and more. Check it out! http://discover.yahoo.com/stayintouch.html

2 1

[Serusers] 0.9.0 version status?
by Jose Bertuzzi 06 Jun '05

06 Jun '05

Hello, is version 0.9.0 stable? Do you recommend to install it, or 0.8.14 works better? I am running 0.8.14 compiled from source on freebsd. I had some trouble trying to compile radius modules so I think 0.9.0 could fix this. thanks for your recommendation, Pablo. --------------------------------- Discover Yahoo! Use Yahoo! to plan a weekend, have fun online & more. Check it out!

2 1

[Serusers] (no subject)
by news 06 Jun '05

06 Jun '05

Hello seruser, I'm using SER 0.8.14 with freeradius 1.0.2 and radiusclient 0.4.8 . All connections are going through a rtpproxy. Now my Question: Is it possible to get a "watchdog" - functionality with SER? I need to get an entry (stop or failure) in my detail-yyyy-mm-dd file when the connection has been terminated by an error or by disconnecting one phone from the net. thanks -- ___________________________ Stefan Abu Salah Stefan(a)AbuSalah.com www.AbuSalah.com 0179 140 89 17

1 0

[serusers] logs
by michael p 06 Jun '05

06 Jun '05

hi where can i find the logs errors and others of ser? thanks M.

2 2

[Serusers] SER + Asterisk
by Siphiwo Dzingwe 06 Jun '05

06 Jun '05

Hi I have recently installed SER 0.8.14 and I am trying to make it talk to asterisk. How do I do that?

1 0

[Serusers] Re: [Serdev] UA request responses that do not conform RFC3261 nor RFC2543
by Jan Janak 06 Jun '05

06 Jun '05

The best way would be to fix the user agent. SER would not know where to forward the reply if it does not contain Via. Jan. On 02-06-2005 10:58, Dominik Schulte wrote: > Hi, > > I am struggling with a problem where I have a UA that responds to invites but > does not preserve the original request Via fields. see ngrep capture below. > I can see that SER receives the reply, but drops it because it cannot correlate any transaction with this response. > I tried to write a module that uses the tm modules callbacks, but it seems that response messages are dropped by the core. > > Is there any way I can validate responses that fall outside any transaction?? > > # > U 172.17.26.19:5060 -> 172.17.26.178:5060 > INVITE sip:dschulte@heat.callsuite.tellumat.co.za SIP/2.0. > Via: SIP/2.0/UDP 172.17.26.19:5060;rport;branch=z9hG4bKE2FD4189B717447BB8192445A851777B. > From: DOMINIKS PC SIP PHONE <sip:1200@heat.callsuite.tellumat.co.za>;tag=859273871. > To: <sip:dschulte@heat.callsuite.tellumat.co.za>. > Contact: <sip:1200@172.17.26.19:5060>. > Call-ID: AF0EF5B2-5850-4573-A2FF-F1D50A6B2B65(a)172.17.26.19. > CSeq: 656 INVITE. > Max-Forwards: 70. > Content-Type: application/sdp. > User-Agent: X-Lite release 1103m. > Content-Length: 293. > > > # > U 172.17.26.178:5060 -> 172.17.26.19:5060 > SIP/2.0 100 trying -- your call is important to us. > Via: SIP/2.0/UDP 172.17.26.19:5060;rport=5060;branch=z9hG4bKE2FD4189B717447BB8192445A851777B. > From: DOMINIKS PC SIP PHONE <sip:1200@heat.callsuite.tellumat.co.za>;tag=859273871. > To: <sip:dschulte@heat.callsuite.tellumat.co.za>. > Call-ID: AF0EF5B2-5850-4573-A2FF-F1D50A6B2B65(a)172.17.26.19. > CSeq: 656 INVITE. > Server: Sip EXpress router (0.8.14 (i386/linux)). > Content-Length: 0. > Warning: 392 172.17.26.178:5060 "Noisy feedback tells: pid=17762 req_src_ip=172.17.26.19 req_src_port=5060 in_uri=sip:dschulte@heat.callsuite.tellumat.co.za out_uri=sip:94099:1001@172.17.1.31 via_cnt==1". > . > > # > U 172.17.26.178:5060 -> 172.17.1.31:5060 > INVITE sip:94099:1001@172.17.1.31 SIP/2.0. > Record-Route: <sip:dschulte@172.17.26.178;ftag=859273871;lr=on>. > Via: SIP/2.0/UDP 172.17.26.178;branch=z9hG4bK33ab.6f0be065.0. > Via: SIP/2.0/UDP 172.17.26.19:5060;rport=5060;branch=z9hG4bKE2FD4189B717447BB8192445A851777B. > From: DOMINIKS PC SIP PHONE <sip:1200@heat.callsuite.tellumat.co.za>;tag=859273871. > To: <sip:94099@172.17.1.31>>. > Contact: <sip:1200@172.17.26.19:5060>. > Call-ID: AF0EF5B2-5850-4573-A2FF-F1D50A6B2B65(a)172.17.26.19. > CSeq: 656 INVITE. > Max-Forwards: 69. > Content-Type: application/sdp. > User-Agent: X-Lite release 1103m. > Content-Length: 293. > > # > U 172.17.1.31:5060 -> 172.17.26.178:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 172.17.26.178. > `From: DOMINIKS PC SIP PHONE <sip:1200@heat.callsuite.tellumat.co.za>;tag=859273871. > To: <sip:94099@172.17.1.31>;tag=viper94099. > Call-ID: AF0EF5B2-5850-4573-A2FF-F1D50A6B2B65(a)172.17.26.19. > CSeq: 656 INVITE. > Content-Length: 0. > . > > > # > U 172.17.1.31:5060 -> 172.17.26.178:5060 > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP 172.17.26.178. > From: DOMINIKS PC SIP PHONE <sip:1200@heat.callsuite.tellumat.co.za>;tag=859273871. > To: "" <sip:94099@172.17.1.31>;tag=viper94099. > Call-ID: AF0EF5B2-5850-4573-A2FF-F1D50A6B2B65(a)172.17.26.19. > CSeq: 656 INVITE. > Content-Length: 0. > . > > > > > # > U 172.17.26.178:5060 -> 172.17.1.31:5060 > INVITE sip:94099:1001@172.17.1.31 SIP/2.0. > Record-Route: <sip:dschulte@172.17.26.178;ftag=859273871;lr=on>. > Via: SIP/2.0/UDP 172.17.26.178;branch=z9hG4bK33ab.6f0be065.0. > Via: SIP/2.0/UDP 172.17.26.19:5060;rport=5060;branch=z9hG4bKE2FD4189B717447BB8192445A851777B. > From: DOMINIKS PC SIP PHONE <sip:1200@heat.callsuite.tellumat.co.za>;tag=859273871. > To: <sip:94099@172.17.1.31>>. > Contact: <sip:1200@172.17.26.19:5060>. > Call-ID: AF0EF5B2-5850-4573-A2FF-F1D50A6B2B65(a)172.17.26.19. > CSeq: 656 INVITE. > Max-Forwards: 69. > Content-Type: application/sdp. > User-Agent: X-Lite release 1103m. > Content-Length: 293. > > > > ********************************************************************** > Relevant company disclaimers are available at the following addresses: > Tellumat (Pty) Ltd e-mail: mailto:disclaimer@tellumat.com?Subject=Tellumat_Disclaimer > Web: http://www.tellumat.com/email.aspx > ********************************************************************** > > _______________________________________________ > Serdev mailing list > serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev

1 0

[Serusers] pb with jabber
by mma 06 Jun '05

06 Jun '05

i add the following entries on the sip_jab.jusers table insert into jusers values ('','user1jabber','password','sip:user1@sip.server.net',0); insert into jusers values ('','user2jabber','password','sip:user1@sip.server.net',0); i created user1jabber and user2jabber using exodus as jabber client i created also user1 and user2 on ser whiwc is running on 192.168.1.70 (sip.server.net) jabber is runninng also on the same machine : 192.168.1.70 (jabber.server.net) jabber works fine, i have tested it; DNS server : 195.167.203.168 i want the two users to communicate using IM user1 is connecting to windows messenger : user1(a)sip.server.net user2 is connecting to exodus : user2jabber(a)jabber.server.net user1 send an IM to user2jabber using user2jabber*jabber.org(a)jabber.server.net the problems are : -user2jabber don't receive what user1 sends. -also pa modules does not work : i have no response from the server when a SUBSCRIBE request is sent (i added user3 as a contact to user1 ) please i need help regards ///////////////////////////////////////////////////////////////////////////////////// No. Time Source Destination Protocol Info 25 12.150821 192.168.1.118 195.167.203.168 DNS Standard query A sip.server.net 26 12.153145 195.167.203.168 192.168.1.118 DNS Standard query response A 192.168.1.70 27 12.160197 192.168.1.118 192.168.1.70 SIP Request: REGISTER sip:sip.server.net 28 12.164436 192.168.1.70 192.168.1.118 SIP Status: 200 OK (1 bindings) 29 12.166088 192.168.1.70 192.168.1.118 SIP Status: 200 Accepted (1 bindings) 30 12.169036 192.168.1.118 192.168.1.70 SIP Request: SUBSCRIBE sip:user3@sip.server.net 31 12.172514 192.168.1.70 192.168.1.118 SIP Status: 400 Bad Request ..... 171 21.841023 192.168.1.118 192.168.1.70 SIP/SDP Request: INVITE sip:user2jabber*jabber.org@jabber.server.net, with session description 172 21.843627 192.168.1.70 192.168.1.118 SIP Status: 202 Accepted 173 21.861278 192.168.1.118 192.168.1.70 SIP Request: ACK sip:user2jabber*jabber.org@jabber.server.net 174 21.862529 192.168.1.118 192.168.1.70 SIP Request: MESSAGE sip:user2abber*jabber.org@jabber.server.net (text/plain) 175 21.865357 192.168.1.70 192.168.1.118 SIP Status: 200 Accepted ..... 297 22.446316 192.168.1.118 192.168.1.70 SIP Request: BYE sip:user2jabber*jabber.org@jabber.server.net

1 2

[Serusers] pb with jabber
by mma 06 Jun '05

06 Jun '05

i add the following entries on the sip_jab.jusers table insert into jusers values ('','user1jabber','password','sip:user1@sip.server.net',0); insert into jusers values ('','user2jabber','password','sip:user1@sip.server.net',0); i created user1jabber and user2jabber using exodus as jabber client i created also user1 and user2 on ser whiwc is running on 192.168.1.70 (sip.server.net) jabber is runninng also on the same machine : 192.168.1.70 (jabber.server.net) jabber works fine, i have tested it; DNS server : 195.167.203.168 i want the two users to communicate using IM user1 is connecting to windows messenger : user1(a)sip.server.net user2 is connecting to exodus : user2jabber(a)jabber.server.net user1 send an IM to user2jabber using user2jabber*jabber.org(a)jabber.server.net the problems are : -user2jabber don't receive what user1 sends. -also pa modules does not work : i have no response from the server when a SUBSCRIBE request is sent (i added user3 as a contact to user1 ) please i need help regards ///////////////////////////////////////////////////////////////////////////////////// No. Time Source Destination Protocol Info 25 12.150821 192.168.1.118 195.167.203.168 DNS Standard query A sip.server.net 26 12.153145 195.167.203.168 192.168.1.118 DNS Standard query response A 192.168.1.70 27 12.160197 192.168.1.118 192.168.1.70 SIP Request: REGISTER sip:sip.server.net 28 12.164436 192.168.1.70 192.168.1.118 SIP Status: 200 OK (1 bindings) 29 12.166088 192.168.1.70 192.168.1.118 SIP Status: 200 Accepted (1 bindings) 30 12.169036 192.168.1.118 192.168.1.70 SIP Request: SUBSCRIBE sip:user3@sip.server.net 31 12.172514 192.168.1.70 192.168.1.118 SIP Status: 400 Bad Request ..... 171 21.841023 192.168.1.118 192.168.1.70 SIP/SDP Request: INVITE sip:user2jabber*jabber.org@jabber.server.net, with session description 172 21.843627 192.168.1.70 192.168.1.118 SIP Status: 202 Accepted 173 21.861278 192.168.1.118 192.168.1.70 SIP Request: ACK sip:user2jabber*jabber.org@jabber.server.net 174 21.862529 192.168.1.118 192.168.1.70 SIP Request: MESSAGE sip:user2abber*jabber.org@jabber.server.net (text/plain) 175 21.865357 192.168.1.70 192.168.1.118 SIP Status: 200 Accepted ..... 297 22.446316 192.168.1.118 192.168.1.70 SIP Request: BYE sip:user2jabber*jabber.org@jabber.server.net ////////////////////////////////////////////////////////////////////////////////////// ser.cfg ***************** debug=4 # debug level (cmd line: -dddddddddd) #fork=yes # (cmd. line: -D) fork=no log_stderror=yes # (cmd line: -E) #log_stderror=no # (cmd line: -E) children=2 check_via=no # (cmd. line: -v) dns=off # (cmd. line: -r) rev_dns=off # (cmd. line: -R) port=5060 listen=sip.server.net listen=192.168.1.70 fifo="/tmp/ser_fifo" #fifo="/tmp/ser_fifoql://ser:heslo@localhost/ser" # for more info: sip_router -h # ------------------ module loading ---------------------------------- #modules loadmodule "/usr/local/lib/ser/modules/print.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/jabber.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/pa.so" # ----------------- setting module-specific parameters --------------- # -- registrar params -- modparam("registrar", "default_expires", 120) modparam("registrar", "use_domain", 1) # -- usrloc params -- modparam("usrloc", "use_domain", 1) modparam("usrloc", "db_mode", 0) # -- jabber params -- modparam("jabber","db_url","mysql://jabber:jabber@127.0.0.1/sip_jab") modparam("jabber","jaddress","jabber.server.net") modparam("jabber","jport",5222) modparam("jabber","workers",2) modparam("jabber","max_jobs",10) modparam("jabber","cache_time",200) modparam("jabber","delay_time",60) modparam("jabber","jdomain","jabber.server.net=*") route{ #if ( !mf_process_maxfwd_header("10") ) #{ # sl_send_reply("483","To Many Hops"); # drop(); #}; if (uri=~"[@:]sip\.server\.net([;:].*)*") #if (uri=~"^sip:(.+@)?(192\.168\.1\.70|(jabber\.)?server\.net)([:;\?].*)?$") { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { if (t_newtran()) { save("location"); log("REGISTER received -> reply okay\n"); }; if(search("registration")) { log("XJAB: Going ONline in Jabber network!!!!\n"); if(jab_go_online()) { sl_send_reply("200", "Accepted"); } else { sl_send_reply("404","Not found"); }; } else { log("XJAB: Going OFFline in Jabber network!!!!\n"); if(jab_go_offline()) { sl_send_reply("200", "Accepted"); } else { sl_send_reply("404","Not found"); }; }; break; }; if (method=="SUBSCRIBE") { log("XJAB:methode SUBSCRIBE \n"); if(jab_go_offline()) if (t_newtran()) { log("XJAB: subscribe registrar\n"); # subscribe("registrar"); handle_subscription("registrar"); }; break; }; if(!lookup("location")) { sl_send_reply("404","Not found"); break; }; }; if (search("To:.*@jabber\.server\.net")) { log("XJAB : search for jabber.sofiphone.net /n"); if (! t_newtran()) { sl_reply_error(); break; }; if (method=="MESSAGE") { log("MESSAGE received -> manage it with XJAB\n"); if(search("\n:on")) { if (jab_go_online()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not found"); }; break; }; if(search("\n:off")) { if (jab_go_offline()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not found"); }; break; }; if(search("\n:join")) { if (jab_join_jconf()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not Found"); }; break; }; if(search("\n:exit")) { if (jab_exit_jconf()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("404","Not Found"); }; break; }; if (jab_send_message()) { sl_send_reply("200","Accepted"); }else{ sl_send_reply("503","Service Unavailable"); }; break; }; if (method=="SUBSCRIBE") { log("subscribe jabber /n"); # subscribe("jabber"); handle_subscription("jabber"); break; }; log("NON_Message request received for JABBER gateway->dropt!\n"); sl_send_reply("202","Accepted"); break; }; if (!t_relay()) { sl_reply_error(); }; #forward(uri:host,uri:port); } /////////////////////////////////////////////////////////////////////////////////////

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[Serusers] SERWEB and MySQL Connection ERROR Database error: pconnect(192.168.1.246, ser, $Password) failed.
by RockV 05 Jun '05

05 Jun '05

Hi, I am not an expert, but I think I have found a solution. First, you should point serweb to your mysql database. //////////////////////////////////////////////////////////////// // configure database /* these are the defaults with which SER installs; if you changed the SER account for MySQl, you need to update here */ $this->db_host="127.0.0.1"; //database host $this->db_name="ser"; //database name $this->db_user="ser"; //database conection user $this->db_pass="heslo"; //database conection password In my case localhost but if I set localhost in $this->db_host it doesn't work. So, you must set IP Address... Then, you should check if you can enter to MySQL with IP Address. In my case: mysql -h 127.0.0.1 -u ser -pheslo ser Is not enough if you can enter with: mysql -h localhost -u ser -pheslo ser I think that will help. RockV >yup password error :-), can you connect to the db from the linux command >line > >mysql -h a.b.c.d -u username -ppassword ser > >do this from the machine serweb is hosted on. > >Also I would avoid serweb until you have ser up and running properly, >because serweb comes with its own little headaches :-) > >Iqbal > >Manoj Ghag wrote: > >> hi, >> In case of MYSQL .. I am getting following ERROR when I try to open a >> SERWEB.... >> >> >> *Warning*: mysql_pconnect(): Access denied for user: 'ser at test' (Using >> password: YES) in */var/www/phplib/db_mysql.inc* on line *73* >> *Database error:* pconnect(192.168.1.246, ser, $Password) failed. >> *MySQL Error*: () >> Session halted. >> >> >> ANY HELP ? >> >> >> ------------------------------------------------------------------------ >> Yahoo! Mail >> Stay connected, organized, and protected. Take the tour >> <HYPERLINK "http://tour.mail.yahoo.com/mailtour.html"http://tour.mail.yahoo.com/mailtou r.html> >> >>------------------------------------------------------------------------ >> >>_______________________________________________ >>Serusers mailing list >>Serusers at iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers >> >> > > > -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.6.2 - Release Date: 04/06/2005

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[Serusers] need help to route calls to PSTN
by harry gaillac 05 Jun '05

05 Jun '05

Hello ser users, I configure ser and asterisk on the same box (+nat). I can send and receive calls from my domain or other domains however i can't send uri to pstn : ser listen on port 5060 / public ip asterisk listen 5040 / private address 192.168.0.50 When I dial a number to pstn phone send sip:1234@mydomain:5060 but ser don't rewrite host port according to ser.cfg . What's wrong ? Why ? # $Id: ser.cfg,v 1.27 2005/03/10 14:16:25 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) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* 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" user=root group=ser fifo_user=root # owner of the ser fifo fifo_group=ser fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) listen=80.119.9.253 # # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/mysql.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/uri.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/permissions.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params-- modparam("usrloc", "db_mode", 2) # -- autdb_params-- modparam("auth_db", "calculate_ha1", yes) 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) # -- nathelper params -- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) # -- registrar params -- modparam("registrar", "nat_flag", 6) # -- tm params -- modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # --domain params -- modparam("domain", "db_mode", 1) # Use caching # -- permissions params -- modparam("permissions", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") # ------------------------- 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 >= max_len ) { 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(); }; if (method=="ACK" || method=="BYE") { unforce_rtp_proxy(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if (has_totag() && method=="INVITE" || method=="ACK") { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # Requests to others domains if (!is_from_local()) { # 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 (is_from_local()) { if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; }; } route[1] { ########################### # Default Message Handler # ########################### t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { ############################ # Register Message Handler # ############################ if (!search("^Contact:") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("nxs.yi.org", "subscriber")) { www_challenge("nxs.yi.org", "0"); sl_send_reply("401", "Unauthorized"); break; }; if (!save("location")) { sl_reply_error(); break; }; } route[3] { ############################# # CANCEL and INVITE handler # ############################# if (nat_uac_test("19")) { setflag(6); }; if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; if (!is_from_local()) { route(1); break; }; if (src_ip==192.168.0.50 && allow_trusted()) { route(4); route(1); } else if (src_ip==192.168.0.50 && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("403", "Forbidden"); break; }; break; }; if (uri=~"sip:^1013@.*$" || uri=~"sip:^[0-9]{10}@.*$" || uri=~"sip:^06[0-9]{8}@.*$") { route(5); route(1); break; }; # native SIP destinations are handled using our USRLOC DB lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); break; } route[4] { ########################## # Incoming PSTN Handler # ########################## if (uri=~"^[0-9]{10}@.*$") { strip(8); break; }; } route[5] { ########################## # Outgoing PSTN Handler # ########################## if (uri=~"sip:^1013@.*$") { if (!is_user_in("Request-URI", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"sip:^[0-9]{2}@.*$") { if (!is_user_in("Request-URI", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"sip:^06[0-9]{8}@.*$") { if (!is_user_in("Request-URI", "cell phone")) { sl_send_reply("403", "Forbidden"); break; }; rewritehostport("192.168.50:5040"); }; } onreply_route[1] { if (isflagset(6) && status=~"(180) | (183) | 2[0-9][0-9]") { force_rtp_proxy(); }; if (nat_uac_test("1")) { fix_nated_contact(); }; } _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

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[Serusers] Early media
by Olivier Taylor 05 Jun '05

05 Jun '05

Hi guys, We use ser, rtpproxy and sems. It seems that some gateways(Cisco) doesn't send the Ringing, only the 100 Trying. Is there a way to force UA to accept early media? It seems that's some UA doesn't accept that early media (Grandstream and Sipura), Eyebeam form Xten works correctly... Any ideas? Olivier -- Olivier Taylor

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[Serusers] SER + SEMS Voicemail Feature
by Narcis GRATIANU 04 Jun '05

04 Jun '05

Hello ! I am using SER with SEMS for voicemail, but, for now, the voicemail is active only when the user is not authenticated. Does anybody has a config sample for the voicemail to kick in after, let's say, few rings, also when the user is authenticated ? Thank you !

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[Serusers] routing to PSTN
by harry gaillac 04 Jun '05

04 Jun '05

Hello, Here is my ser.cfg I try to send uri not in location or aliase table . INVITE message is not sent to gateway just 404 is returned ? What's wrong ? Harry # $Id: ser.cfg,v 1.27 2005/03/10 14:16:25 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) #memlog=5 # memory debug log level #log_facility=LOG_LOCAL0 # sets the facility used for logging (see syslog(3)) /* 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" user=root group=ser fifo_user=root # owner of the ser fifo fifo_group=ser fifo_mode=0660 # fifo's permissions #disable_core=yes #disables core dumping #open_fd_limit=1024 # sets the open file descriptors limit #mhomed=yes # usefull for multihomed hosts, small performance penalty #disable_tcp=yes #tcp_accept_aliases=yes # accepts the tcp alias via option (see NEWS) listen=80.119.10.96 alias="nxs.yi.org" # # ------------------ module loading ---------------------------------- loadmodule "/usr/lib/ser/modules/mysql.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/uri.so" loadmodule "/usr/lib/ser/modules/nathelper.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/permissions.so" loadmodule "/usr/lib/ser/modules/group.so" loadmodule "/usr/lib/ser/modules/domain.so" loadmodule "/usr/lib/ser/modules/avpops.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params-- modparam("usrloc", "db_mode", 2) # -- autdb_params-- modparam("auth_db", "calculate_ha1", yes) 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) # -- nathelper params -- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) # -- registrar params -- modparam("registrar", "nat_flag", 6) # -- tm params -- modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") # -- permissions params -- modparam("permissions", "db_url", "mysql://ser:heslo@localhost/ser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("permissions", "default_deny_file", "/etc/ser/permissions.deny") modparam("permissions", "default_allow_file", "/etc/ser/permissions.allow") # ------------------------- 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 >= max_len ) { 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(); }; if (method=="ACK" || method=="BYE") { unforce_rtp_proxy(); }; # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { if (has_totag() && method=="INVITE" || method=="ACK") { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # Requests to others domains 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=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; # native SIP destinations are handled using our USRLOC DB lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); }; } route[1] { ########################### # Default Message Handler # ########################### t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { ############################ # Register Message Handler # ############################ if (!search("^Contact:") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("nxs.yi.org", "subscriber")) { www_challenge("nxs.yi.org", "0"); sl_send_reply("401", "Unauthorized"); break; }; if (!save("location")) { sl_reply_error(); break; }; } route[3] { ############################# # CANCEL and INVITE handler # ############################# if (nat_uac_test("19")) { setflag(6); }; if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; if (!uri==myself) { route(1); break; }; if (src_ip==192.168.0.50 && allow_trusted()) { route(4); } else if (src_ip==192.168.0.50 && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); sl_send_reply("403", "Forbidden"); break; }; break; }; if (uri=~"sip:^[0-9]{2}@.*$" || uri=~"sip:^1013@.*$" ||uri=~"sip:^[0-9]{10}@.*$" || uri=~"sip:^06[0-9]{8}@.*$") { route(5); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; t_on_reply("1"); if (!t_relay()) { if (isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[4] { ########################## # Incoming PSTN Handler # ########################## if (uri=~"^[0-9]{10}@.*$") { strip(8); break; }; lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[5] { ########################## # Outgoing PSTN Handler # ########################## if (uri=~"sip:^[0-9]{2}@.*$") { if (!is_user_in("Request-URI", "internal")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"sip:^[0-9]{2}@.*$") { if (!is_user_in("Request-URI", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"sip:^1013@.*$") { if (!is_user_in("Request-URI", "local")) { sl_send_reply("403", "Forbidden"); break; }; } else if (uri=~"sip:^06[0-9]{8}@.*$") { if (!is_user_in("Request-URI", "cell phone")) { sl_send_reply("403", "Forbidden"); break; }; }; rewritehostport("192.168.0.50:5040"); route(1); } onreply_route[1] { if (isflagset(6) && status=~"(180) | (183) | 2[0-9][0-9]") { force_rtp_proxy(); }; if (nat_uac_test("1")) { fix_nated_contact(); }; } _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

1 0

[Serusers] billing question (some architectural case)
by Wojciech Ziniewicz 04 Jun '05

04 Jun '05

Hello, I 've just came to the point of rolling my own billing that I have mysql database with table "acc" and details of called party , accstart, acc period and so on. I have another table that I have prefixes and prices for T1,T2,T3 . My question is : do You have some howtos, scripts for rating users? I try bash for now but it is slow and not scalable in future.. what about mysql shell or sth? Could you send me some examples ? I would appreciate that Regards..... -- Pozdrawiam, Wojciech Ziniewicz Optocomp sp.z.o.o, www.optocomp.pl mailto: wojtekz(a)optocomp.pl +48(0)691031535

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[Serusers] ser don't start
by harry gaillac 04 Jun '05

04 Jun '05

Hello ser users, I've sent many mails about starting ser. my problem is solved . ser sent to syslog: auth_db:str_fixup: Invalid table version (use ser_mysql.sh reinstall) auth_db module did not find table "subsciber" below. if (!www_authorize("nxs.yi.org", "subsciber")) { www_challenge("nxs.yi.org", "0"); sl_send_reply("401", "Unauthorized"); break; }; I fix table to "subscriber" ! Thanks ser users for help Regards Harry _____________________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 1 Go d'espace de stockage pour vos mails, photos et vidéos ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com

1 0

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