- sr-users - kamailio.org

by Daniel-Constantin MIERLA

Hello, you must have 'expat' library installed ('expat' and 'expat-devel' packages) -- it is an XML parser very common in unix/linux distributions. Best regards, Daniel On 5/6/2003 2:02 AM, Phil Yuska wrote: >After watching this a while longer, it looks as though the session drops >are being caused by transport failures on the jabber server (aim and or >yahoo). It was pretty stable with out them loaded. > >I downloaded and compiled the latest cvs snapshot but the jabber.so >module won't load, it gets this error: > >0(126) ERROR: load_module: could not open module ></usr/local/lib/ser/modules/jabber.so>: >/usr/local/lib/ser/modules/jabber.so: Undefined symbol >"XML_GetCurrentByte >Index" > >Phil > > > >-----Original Message----- >From: serusers-admin(a)iptel.org [mailto:serusers-admin@lists.iptel.org] On >Behalf Of Daniel-Constantin MIERLA >Sent: Saturday, May 03, 2003 3:54 PM >To: serusers(a)lists.iptel.org >Subject: Re: [Serusers] jabber module > >Hello, > >On 5/3/2003 12:47 AM, Phil Yuska wrote: > > > >>Are the sessions between the jabber.so module and the jabber server >>persistent? >> >> >> >not really. There is a parameter (cache_time) to set how long a >connection to Jabber server is kept alive if there is no traffic through >it. > > > >>It seems like the session establishes then drops after each >>message is delivered to the jabber server resulting in this error being >>reported back to the client. >> >> >> >Could you capture the network traffic between ser and jabber server -- >using ngrep or ethereal? > > > >>ERROR: Connection to Jabber server lost. You have to login to Jabber >>server again (join the conferences again that you were participating, >>too) >>sip_to_jabber_gateway says: >>INFO: Your are now offline in Jabber network. >> >> >> >These messages are sent when the connection between jabber gateway and >jabber server gets down without reason. > > > >>I'm also seeing these errors, but I'm not sure if they indicate a >>problem or if it's just house keeping done by ser. >> >>1(11705) XJAB:xjab_check_workers: error - worker[0][pid=11717] lost >>forever >>1(11705) XJAB:xjab_check_workers: worker[1][pid=11719] has exited - >>status=0 err=-1 errno=10 >> >> >> >here somehow a jabber worker dies. Please try again and see if you can >generate a core (use "ulimit -c unlimited"). If you get one make an >archive with sources, logs, binary files and core file, make them >available on a ftp/http server or send them to me or to >serhelp(a)lists.iptel.org. Before doing that, you may also try the latest CVS >snapshot if you use an old version - it could be an already fixed bug. > > > >>Finally can the presence module be used with version 0.8.10? >> >> >> >No, the presence agent is not available in 0.8.10 as well as its >integration with jabber module. Next release will include both of them. > >Best regards, >Daniel > > > >>Regards, >> >>Phil >> >> >>_______________________________________________ >>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 > > > > > >

21 years, 2 months

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RE: [Serusers] Firewall

by jaime.gill＠orange.co.uk

Hello, If you are really interested in having SER within a natted network or running on the firewall/nat itself, may be you could give a try to the fcp module. It relies on a client side which is added as a module to SER, and a server side, running on the firewall/nat (with iptables). The module keeps track of sessions similar to a b2bua. When a new request for a session comes (INVITE, SUBSCRIBE, MESSAGE, etc.) from an internal client, the fcp module learns the external IP address and a port on the firewall and makes several changes to the SIP message. In the current implementation, Contact and SDP can be changed before sending any request through the firewall/nat. When responses come back (200 OK with SDP), the firewall ports are open for media to flow. Ports are closed after expiration of rules or because of CANCEL/BYE are issued from any of the end points. This has been tested so far in the following scenario: SIP UA1 ----- SER+fcp module ------ NAT/FW(fcpd) --------- SER ----------- SIP UA2 With the current version of fcpd (http://www.iptel.org/fcp/) I have not been successful in establishing a media connection, but you might be luckier :) However, the previous version worked for me in several occasions (I could hear audio to and from SIP UA1/SIP UA2). If your are interested in giving it a try, let me know and we see how far we get. Jaime "Hans Scheffers" <hans.scheffers(a)xs4all.nl> on 06/05/2003 13:32:16 To: serusers(a)lists.iptel.org cc: (bcc: Jaime GILL/EN/HTLUK) Subject: RE: [Serusers] Firewall NAT, i have one public ip The problem with iptable/ipchains is the way they filter compared to Cisco a.s.o. Hans Scheffers JifLin B.V. Leliestraat 7 7151 GH Eibergen http://www.jiflin.nl > -----Oorspronkelijk bericht----- > Van: Jan Janak [mailto:jan@iptel.org] > Verzonden: dinsdag 6 mei 2003 12:18 > Aan: Hans Scheffers > CC: serusers(a)lists.iptel.org > Onderwerp: Re: [Serusers] Firewall > > > BTW, are you behind a NAT or just a firewall ? > > Jan. > > On 06-05 11:36, Hans Scheffers wrote: > > But are there developers working on it? > > > > > > Hans Scheffers > > JifLin B.V. > > Leliestraat 7 > > 7151 GH Eibergen > > > > http://www.jiflin.nl > > > > > > > -----Oorspronkelijk bericht----- > > > Van: Jan Janak [mailto:jan@iptel.org] > > > Verzonden: dinsdag 6 mei 2003 11:18 > > > Aan: Juha Heinanen > > > CC: Hans Scheffers; serusers(a)lists.iptel.org > > > Onderwerp: Re: [Serusers] Firewall > > > > > > > > > On 06-05 07:54, Juha Heinanen wrote: > > > > Jan Janak writes: > > > > > > > > > > I have an Astaro Linux Firewall. This firewall blocks > > > everything (what I > > > > > > want :)), and is based on on iptables. > > > > > > > > if it based on iptables, then the right solution is to > write a sip > > > > helper application for iptables. everything else is hackery. > > > > > > And this is very tricky, that is the reason why there is no such > > > helper application yet. > > > > > > Jan. > > > > > > > > > > _______________________________________________ > > 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 ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. *******************************************************************************

21 years, 2 months

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[Serusers] Forward when unavailable

by radan＠nasty.gda.pl

I have problem with forwarding unanswered calls after fr_inv_timer and fr_timer parameters I do ( I hope so) everything according with manual but it doesn't work :( My configuration file: # tm -parametry modparam("tm", "fr_inv_timer", 8) modparam("tm", "fr_timer", 5) # ------------------------- request routing logic ------------------- # main routing logic alias="gda.pl" alias="sips.gda.pl" route{ # initial sanity checks -- messages with # max_forwars==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (len_gt( max_len )) { sl_send_reply("513", "Message too big"); break; }; # Do strict routing if pre-loaded route headers present rewriteFromRoute(); # 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("gda.pl", "subscriber")) { www_challenge("gda.pl", "0"); break; }; save("location"); log(3,"REGISTER zarejestrowany uzytkownik radan"); sl_send_reply("200", "ok"); break; }; lookup("aliases"); # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; if (uri=~"^sip:radan@gda\.pl") { seturi("sip:unknown@gda.pl"); |------ should be this addresses exist ?? append_branch("sip:nobody@gda.pl:9"); |------ t_on_negative("1"); t_relay(); }; # forward to current uri now if (!t_relay()) { sl_reply_error(); }; } reply_route[1] { append_branch("sip:2222@gda.pl"); - on this addres I want to redirect unanswered call log(3,"przekierowanie"); } ------------------END OF FILE--------------------- Andrzej Radke

21 years, 2 months

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[Serusers] Strict routing

by Gulherme Dal Pizzol

Fellows, I started using the cvs version of ser instead of 0.8.10. But I'm facing some doubts on strict routing vs. loose routing. In 0.8.10 I was using: addRecordRoute(); rewriteFromRoute(); Now in CVS version, by default strict routing is disabled, right? So, according to the rr module, I need to use record_route_strict, so I compiled rr module with it. Now, I'm using record_route_strict() instead of addRecordRoute() and loose_route() instead of rewriteFromRoute(); Is this correct, or am I missing something? Thanks in advance, Guilherme.

21 years, 2 months

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[Serusers] nathelper module in cvs updated

by Maxim Sobolev

Folks, I've just finished merging changes from our development version of nathelper into the ser's cvs. The following changes were made: - Don't apply contact address rewriting in SDP body if the original address is 0.0.0.0, which usially present in re-INVITEs and means that sender asks to temporarly suspend media session; - save original address into the oldmediaip option in the SDP body, which is useful for debugging; - fix contact URI parsing routine to understand those URIs in the form sip:<ip>:<port>, which is the case with URIs generated by the Windows Messenger; - some style fixes and general diff reduction with our development version. All nathelper users are encouraged to test it and report any problems to me, since it is probably the version that will be delivered with the next ser release. Thanks! -Maxim

21 years, 2 months

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[Serusers] jabber module

by Phil Yuska

Are the sessions between the jabber.so module and the jabber server persistent? It seems like the session establishes then drops after each message is delivered to the jabber server resulting in this error being reported back to the client. ERROR: Connection to Jabber server lost. You have to login to Jabber server again (join the conferences again that you were participating, too) sip_to_jabber_gateway says: INFO: Your are now offline in Jabber network. I'm also seeing these errors, but I'm not sure if they indicate a problem or if it's just house keeping done by ser. 1(11705) XJAB:xjab_check_workers: error - worker[0][pid=11717] lost forever 1(11705) XJAB:xjab_check_workers: worker[1][pid=11719] has exited - status=0 err=-1 errno=10 Finally can the presence module be used with version 0.8.10? Regards, Phil

21 years, 2 months

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[Serusers] SYSLOG Error Messages

by Ricardo Villa

Hi, We have deployed many Cisco ATAs behind NAT devices. In order to keep the NAT session binding alive, we enabled a feature on the ATAs that basically send a small dummy packet to SER every 90 seconds. Everything works great. The question is, how can I suppress the WARNING/ERRORS that these dummy packets present to the SYSLOG? Is there a simple way to do this in the ser.cfg file? These are the SYSLOG messages: May 1 09:20:58 maui /usr/sbin/ser[23389]: WARNING: upstream bug - 0-terminated packet May 1 09:20:58 maui /usr/sbin/ser[23389]: ERROR: parse_first_line: message too short: 3 May 1 09:20:58 maui /usr/sbin/ser[23389]: ERROR:parse_first_line: bad message May 1 09:20:58 maui /usr/sbin/ser[23389]: ERROR: parse_msg: message=<> May 1 09:20:58 maui /usr/sbin/ser[23389]: ERROR: receive_msg: parse_msg failed Thanks, Ricardo

21 years, 2 months

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RE: [Serusers] SYSLOG Error Messages

by Dinesh

We have been deploying ATA 186s using port forwarding. Am I correct in understanding that you have been deploying with NAT but without the need for port forwarding. Could you please describe the required settings on the ATA Thanks, Dinesh -----Original Message----- From: "Ricardo Villa"<ricvil(a)epm.net.co> Sent: 01-May-03 10:35:58 PM To: "serusers(a)lists.iptel.org"<serusers(a)lists.iptel.org> Subject: [Serusers] SYSLOG Error Messages Hi, We have deployed many Cisco ATAs behind NAT devices. In order to keep the NAT session binding alive, we enabled a feature on the ATAs that basically send a small dummy packet to SER every 90 seconds. Everything works great. The question is, how can I suppress the WARNING/ERRORS that these dummy packets present to the SYSLOG? Is there a simple way to do this in the ser.cfg file? These are the SYSLOG messages: May 1 09:20:58 maui /usr/sbin/ser[23389]: WARNING: upstream bug - 0-terminated packet May 1 09:20:58 maui /usr/sbin/ser[23389]: ERROR: parse_first_line: message too short: 3 May 1 09:20:58 maui /usr/sbin/ser[23389]: ERROR:parse_first_line: bad message May 1 09:20:58 maui /usr/sbin/ser[23389]: ERROR: parse_msg: message=<> May 1 09:20:58 maui /usr/sbin/ser[23389]: ERROR: receive_msg: parse_msg failed Thanks, Ricardo

21 years, 2 months

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[Serusers] presence agent works again

by Jan Janak

Hello, since several people asked: the presence agent module in the CVS compiles and works again. Jan.

21 years, 2 months

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[Serusers] using aliases

by Phil Yuska

Howdy, I am trying to determine why a call won't connect when dialed using an alias. I'm running SIP Express Router v0.8.10 on FreeBSD 4.8-RELEASE. Below is the message the client (Xten X-Lite) is receiving and apparently discarding. It appears as though the call is being routed but I'm puzzled by these message headers. RECEIVE << 192.168.1.11:5060 INVITE sip:pyuska@192.168.1.25:5060 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 192.168.1.11;branch=z9hG4bKad62.cf628db7.0 Via: SIP/2.0/UDP 192.168.1.15:5060 From: yuska <sip:pyuska-ppc@shitworks.com>;tag=585911808 To: <sip:5315@shitworks.com> Contact: <sip:pyuska-ppc@192.168.1.15:5060> Call-ID: 06E1E114-22A1-8B9D-2891-72E17E9928EA(a)192.168.1.15 CSeq: 12625 INVITE Content-Type: application/sdp Content-Length: 283 Do I need to configure the SER server re-write the To: address or is the client misbehaving by not responding to the Contact: ? Any light you could shed on this would be appreciated. Phil

21 years, 2 months

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Re: [Serusers] information about registered users

by Jiri Kuthan

At 12:14 PM 4/30/2003, radan(a)nasty.gda.pl wrote: >Hello all ! >I'm new user of ser :) > >I'd like have a information about registered users in my syslog > > save("location"); > log(3,"REGISTER zarejestrowany uzytkownik $USER"); > sl_send_reply("200", "ok"); ^^^^^^ > break; > }; > > >is some variable to use ? currently not. We are planning it for the future. -Jiri

21 years, 2 months

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[Serusers] information about registered users

by radan＠nasty.gda.pl

Hello all ! I'm new user of ser :) I'd like have a information about registered users in my syslog save("location"); log(3,"REGISTER zarejestrowany uzytkownik $USER"); sl_send_reply("200", "ok"); ^^^^^^ break; }; is some variable to use ? Andrzej Radke sip:radan@task.gda.pl

21 years, 2 months

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[Serusers] How to calculate new HASH_TABLE_SIZE value when extending parser?

by Maxim Sobolev

Hi, I am currently trying to extend ser's parser to allow it recognize User-Agent headers, but stuck into the magic HASH_TABLE_SIZE value. Is there any algorithm to calculate it when adding support for a new header type? Maybe there is some documentation, apart from comments in the code itself, on extending parser, or at least on its inner details? If so, it would be nice to obtain it if possible. Thanks! -Maxim

21 years, 2 months

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[Serusers] Extending usrloc module to allow to save User-Agent values [patch for review]

by Maxim Sobolev

Folks, Attached please find a patch, which extends usrloc/registrar modules to save values from User-Agent field in REGISTER messages into the database. It would be nice to have it included into the next release. Please disregard hackish detection of User-Agent fields, I am currently working on extending parser to recognize that type of field. I'll post updated patch when finished. Also, there is a new unrelated function which can be used to retrieve full list of all currently registered users in all domains. I am using this function in new version of my nathelper modules to do periodical UDP "pinging" of all registered users to keep their NAT bindings alive. I would like to have it reviewed and included into the next release as well. Thanks! -Maxim

21 years, 2 months

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Re: [Serusers] Load Balancing

by Jiri Kuthan

At 03:33 PM 4/22/2003, Alejandro Olchik wrote: >I need some help to implement load balance >when sending invites to an external sip >proxy server. > >I would like to be able to balance load >between to IPs (10.0.0.1 and 10.0.0.2) and >use the backup IP when the primary one >fails. > >Below is the routing code I have: > > > if (!lookup("location")) { > rewritehost("10.0.0.1"); > if (!t_relay_to("10.0.0.1","5060")) { > sl_send_reply("404", "Not Found"); > }; > break; > }; > >How can I add this behaviour? That depends on the distribution scheme you would like to use. For example, you can implement a distribution function that splits requests based on some downstream weights. e.g., modparam("distrib", "weights", # ... # rewrite host-part with one of values using the distribution # 20-30-50 distribute("10.0.0.1/20%, 10.0.0.2/30%, 10.0.0.3/50%) t_on_negative("1") t_relay_to(); # ... reply_route[1] { # look at destinations that failed, and update their probabilities # according to some strategy; for example, temporary 0% weight -- # note that this would take shared memory for keeping the weights # and a timer update_weights(); # if coupled to TM, it can look at previous attempts to eliminate # retrying to a previously failed destination distribute("10.0.0.1/20%, 10.0.0.2/30%, 10.0.0.3/50%) } -Jiri

21 years, 2 months

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RE: [Serusers] problem with cisco 2600 to pstn

by Dan Austin

On your pots dial-peer set the codec to g711alaw, or configure your SIP clients to use that codec. Dan -----Original Message----- From: Yang Xiang [mailto:yang.xiang@iitb.fraunhofer.de] Sent: Friday, April 25, 2003 7:55 AM To: serusers(a)lists.iptel.org Subject: [Serusers] problem with cisco 2600 to pstn Hi all, I am expericing problem with the cisco 2600, which should function as the sip2pstn gateway. If I try to complete a call from a sip phone to pstn, the router says: ------------------------------------------------------------------------ 00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref = 0x03 00:15:49: Bearer Capability i = 0x8090A2 ^^^^^^ 00:15:49: Channel ID i = 0x83 00:15:49: Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have in-band info 00:15:49: Calling Party Number i = 0x80, 'yang2', Plan:Unknown, Type:Unknown 00:15:49: Called Party Number i = 0xC9, '6091574', Plan:Private, Type:Subscriber(local) 00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8 callref = 0x83 00:15:49: Channel ID i = 0x89 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available 00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8 callref = 0x83 00:15:49: Cause i = 0x80C1 - Bearer capability not implemented ^^^^^^^^^^^^^^^^^^^^^^^^^^ 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available ------------------------------------------------------------------------ ---- ------------ Please notice the line "Bearer Capability i = 0x8090A2", the digits "80" mean that this is a ITU voice call, "90" mean circuit mode, 64 kbps and "A2" is for G.711 u-law. So if I call the router from a normal telephone, the debugging looks as follows: ------------------------------------------------------------------------ ---- ---- 01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref = 0x01 01:01:58: Bearer Capability i = 0x8090A3 ^^^ 01:01:58: Channel ID i = 0x89 01:01:58: Calling Party Number i = 0x0181, '609157', Plan:ISDN, Type:Unknown 01:01:58: Called Party Number i = 0xC1, '20', Plan:ISDN, Type:Subscriber(local) 01:01:58: High Layer Compat i = 0x9181 01:01:58: ISDN BR1/0: Event: Received a VOICE call from 609157 on B1 at 64 Kb/s ------------------------------------------------------------------------ ---- --------- whereat the bearer capability is "0x8090A3". It means that the ISDN-switch of German Telekom uses G.711 a-law. I am afraid that is the reason why the sip-call doesn't go through. But I can't find any way to configure this. Has anybody in this mailinglist the same experience? Any hints would be very appreciated. Thanks yang _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

21 years, 2 months

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Re: [Serusers] problem with cisco 2600 to pstn

by Yang Xiang

Hi Jiri, it works! Thank you very much! The voice-port block is the key: ------------------------------- voice-port 1/0/0 compand-type a-law cptone DE bearer-cap Speech ------------------------------ yang ----- Original Message ----- From: "Jiri Kuthan" <jiri(a)iptel.org> To: "Yang Xiang" <yang.xiang(a)iitb.fraunhofer.de> Sent: Friday, April 25, 2003 5:21 PM Subject: Re: [Serusers] problem with cisco 2600 to pstn > try to look at our config at http://www.iptel.org/~jiri/etc/cisco/ios_2003.txt > if it helps you. I think we had the same problem, changed some settings and it > worked then. I unfornantely don't remember what it was. > > -Jiri > > At 04:54 PM 4/25/2003, you wrote: > >Hi all, > > > >I am expericing problem with the cisco 2600, which should function as the > >sip2pstn gateway. If I try to complete a call from a sip phone to pstn, the > >router says: > >------------------------------------------------------------------------ > > 00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref = 0x03 > >00:15:49: Bearer Capability i = 0x8090A2 > > ^^^^^^ > >00:15:49: Channel ID i = 0x83 > >00:15:49: Progress Ind i = 0x8181 - Call not end-to-end ISDN, may > >have in-band info > >00:15:49: Calling Party Number i = 0x80, 'yang2', Plan:Unknown, > >Type:Unknown > >00:15:49: Called Party Number i = 0xC9, '6091574', Plan:Private, > >Type:Subscriber(local) > >00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8 callref = 0x83 > >00:15:49: Channel ID i = 0x89 > >00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now > >available > >00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8 callref = 0x83 > >00:15:49: Cause i = 0x80C1 - Bearer capability not implemented > > > >^^^^^^^^^^^^^^^^^^^^^^^^^^ > >00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now > >available > >--------------------------------------------------------------------------- - > >------------ > > > >Please notice the line "Bearer Capability i = 0x8090A2", the digits "80" > >mean that this is a ITU voice call, "90" mean circuit mode, 64 kbps and "A2" > >is for G.711 u-law. > > > >So if I call the router from a normal telephone, the debugging looks as > >follows: > >--------------------------------------------------------------------------- - > >---- > >01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref = 0x01 > >01:01:58: Bearer Capability i = 0x8090A3 > > ^^^ > >01:01:58: Channel ID i = 0x89 > >01:01:58: Calling Party Number i = 0x0181, '609157', Plan:ISDN, > >Type:Unknown > >01:01:58: Called Party Number i = 0xC1, '20', Plan:ISDN, > >Type:Subscriber(local) > >01:01:58: High Layer Compat i = 0x9181 > >01:01:58: ISDN BR1/0: Event: Received a VOICE call from 609157 on B1 at 64 > >Kb/s > >--------------------------------------------------------------------------- - > >--------- > > > >whereat the bearer capability is "0x8090A3". It means that the ISDN-switch > >of German Telekom uses G.711 a-law. > > > >I am afraid that is the reason why the sip-call doesn't go through. But I > >can't find any way to configure this. > > > >Has anybody in this mailinglist the same experience? > > > >Any hints would be very appreciated. > > > >Thanks > > > >yang > > > > > > > > > > > > > >_______________________________________________ > >Serusers mailing list > >serusers(a)lists.iptel.org > >http://lists.iptel.org/mailman/listinfo/serusers > > -- > Jiri Kuthan http://iptel.org/~jiri/ > >

21 years, 2 months

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[Serusers] SERAdmin program in development to benefit all SER administrators

by Xten India

Hi , Xten Networks, Inc. (www.xten.com) is pleased to announce it is making a contribution to the SER project. The Xten SER Team is comprised of 2 senior engineers and 1 project manager who are committed full-time to the development of SERAdmin.SERAdmin is a GUI interface between SIP Express Router (SER) and the SER administrator. Project location (http://developer.berlios.de/projects/seradmin/) SERAdmin provides control over many SER tasks such as: start, stop, pause, re-start, monitor, add user, edit user, etc. SERAdmin has an intuitive look and feel.SERAdmin is open source, is being developed to benefit all SER administrators, and the feature set of SERAdmin will be determined by the iptel.org SER users' group.So please communicate with the Xten SER Team, post your comments in the public forums, and make use of the Xten SER Team as they are working for the SER community. About Xten (www.xten.com) Xten Networks, Inc. is a leading provider of high-quality SIP Voice over Internet Protocol (VoIP) software. Xten provides IP Telephony products directly to end users, the Enterprise market, Next-Gen Service Providers (ITSPs & Tier 2), Wireless Internet Service Providers (WISPs), Telephone Companies (TELCOs), and Original Equipment Manufacturers (OEMs). Regards,Xten Team --------------------------------- Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo.

21 years, 2 months

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[Serusers] problem with cisco 2600 to pstn

by Yang Xiang

Hi all, I am expericing problem with the cisco 2600, which should function as the sip2pstn gateway. If I try to complete a call from a sip phone to pstn, the router says: ------------------------------------------------------------------------ 00:15:49: ISDN BR1/0: TX -> SETUP pd = 8 callref = 0x03 00:15:49: Bearer Capability i = 0x8090A2 ^^^^^^ 00:15:49: Channel ID i = 0x83 00:15:49: Progress Ind i = 0x8181 - Call not end-to-end ISDN, may have in-band info 00:15:49: Calling Party Number i = 0x80, 'yang2', Plan:Unknown, Type:Unknown 00:15:49: Called Party Number i = 0xC9, '6091574', Plan:Private, Type:Subscriber(local) 00:15:49: ISDN BR1/0: RX <- SETUP_ACK pd = 8 callref = 0x83 00:15:49: Channel ID i = 0x89 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available 00:15:49: ISDN BR1/0: RX <- DISCONNECT pd = 8 callref = 0x83 00:15:49: Cause i = 0x80C1 - Bearer capability not implemented ^^^^^^^^^^^^^^^^^^^^^^^^^^ 00:15:49: Progress Ind i = 0x8188 - In-band info or appropriate now available ---------------------------------------------------------------------------- ------------ Please notice the line "Bearer Capability i = 0x8090A2", the digits "80" mean that this is a ITU voice call, "90" mean circuit mode, 64 kbps and "A2" is for G.711 u-law. So if I call the router from a normal telephone, the debugging looks as follows: ---------------------------------------------------------------------------- ---- 01:01:58: ISDN BR1/0: RX <- SETUP pd = 8 callref = 0x01 01:01:58: Bearer Capability i = 0x8090A3 ^^^ 01:01:58: Channel ID i = 0x89 01:01:58: Calling Party Number i = 0x0181, '609157', Plan:ISDN, Type:Unknown 01:01:58: Called Party Number i = 0xC1, '20', Plan:ISDN, Type:Subscriber(local) 01:01:58: High Layer Compat i = 0x9181 01:01:58: ISDN BR1/0: Event: Received a VOICE call from 609157 on B1 at 64 Kb/s ---------------------------------------------------------------------------- --------- whereat the bearer capability is "0x8090A3". It means that the ISDN-switch of German Telekom uses G.711 a-law. I am afraid that is the reason why the sip-call doesn't go through. But I can't find any way to configure this. Has anybody in this mailinglist the same experience? Any hints would be very appreciated. Thanks yang

21 years, 2 months

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[Serusers] msg's not reaching all the way

by magnus melin

hey all! We´re struggeling with a problem since some time now. We hade set up an SER - jabber environment, the applications are running on differnt servers. SIP-client -------(sip (udp) ) --------> SER-server -----------> Jabber-server ------(xml (tcp) ) -------> Jabber-client <------------------------------ <---------- <---- <------- Now to the problem. There is no problem with registration on either sides, and one can send a msg from the SIP- client to jabber-client, but when responding from jabber to SIP, the SER sends two MESSAGE to the sip-client and get two 400 BAD REQUEST (frame 11 & 12 ) back from the client, and then the jabber-client goes "away" (frame 19) (losses the tcp-handle ?) In short: a message is sent from the sipclient to the jabberclient, the response from the jabberclient wont display on the sip-client. Though it looks like in etherreal that the msg reaches all the way to the sip-client. We have both loose_route() and record_route() (see ser.cfg_jabber attached). Thank's Best Reg Anna & Magnus Frame 7 (520 on wire, 520 captured) Frame Number: 7 Linux cooked capture Packet type: Unicast to us (0) Source: <MISSING> Protocol: IP (0x0800) Internet Protocol, Src Addr: dhcp164-devel (172.18.3.164), Dst Addr: dhcp164-devel (172.18.3.164) Version: 4 Total Length: 504 Time to live: 64 Protocol: UDP (0x11) Source: dhcp164-devel (172.18.3.164) Destination: dhcp164-devel (172.18.3.164) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 484 Checksum: 0xb971 (correct) Session Initiation Protocol Request line: MESSAGE sip:x@e-horizon.se SIP/2.0 Message Header Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bKa858.b1ed4bd6.0 To: sip:x@e-horizon.se From: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se>;tag=949cc72a86e24d5dcb2bf51338531b31-4809 CSeq: 10 MESSAGE Call-ID: 38231eab-9474(a)172.18.3.164 Content-Length: 20 User-Agent: Sip EXpress router(0.8.11pre13 (i386/linux)) Content-Type: text/plain Contact: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se> Session Description Protocol Invalid line: bbbbbbbbbbbbbbbbbbbb Frame 8 (674 on wire, 674 captured) Linux cooked capture Packet type: Sent by us (4) Link-layer address type: 1 Link-layer address length: 6 Protocol: IP (0x0800) Internet Protocol, Src Addr: dhcp164-devel (172.18.3.164), Dst Addr: shot.e-horizon.se (172.18.1.66) Time to live: 64 Protocol: UDP (0x11) Header checksum: 0xdb50 (correct) Source: dhcp164-devel (172.18.3.164) Destination: shot.e-horizon.se (172.18.1.66) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 7203 (7203) Source port: 5060 (5060) Destination port: 7203 (7203) Length: 638 Checksum: 0x6016 (correct) Session Initiation Protocol Request line: MESSAGE sip:172.18.1.66:7203 SIP/2.0 Message Header Record-Route: <sip:x@172.18.3.164:5060;ftag=949cc72a86e24d5dcb2bf51338531b31-4809;lr> Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bK8984.7c66ebe.0 Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bKa858.b1ed4bd6.0 To: sip:x@e-horizon.se From: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se>;tag=949cc72a86e24d5dcb2bf51338531b31-4809 CSeq: 10 MESSAGE Call-ID: 38231eab-9474(a)172.18.3.164 Content-Length: 20 User-Agent: Sip EXpress router(0.8.11pre13 (i386/linux)) Content-Type: text/plain Contact: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se> Session Description Protocol Invalid line: bbbbbbbbbbbbbbbbbbbb Frame 9 (520 on wire, 520 captured) Linux cooked capture Source: <MISSING> Protocol: IP (0x0800) Internet Protocol, Src Addr: dhcp164-devel (172.18.3.164), Dst Addr: dhcp164-devel (172.18.3.164) Protocol: UDP (0x11) Header checksum: 0xd988 (correct) Source: dhcp164-devel (172.18.3.164) Destination: dhcp164-devel (172.18.3.164) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 484 Session Initiation Protocol Request line: MESSAGE sip:x@e-horizon.se SIP/2.0 Message Header Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bKa858.b1ed4bd6.0 To: sip:x@e-horizon.se From: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se>;tag=949cc72a86e24d5dcb2bf51338531b31-4809 CSeq: 10 MESSAGE Call-ID: 38231eab-9474(a)172.18.3.164 Content-Length: 20 User-Agent: Sip EXpress router(0.8.11pre13 (i386/linux)) Content-Type: text/plain Contact: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se> Session Description Protocol Invalid line: bbbbbbbbbbbbbbbbbbbb Frame 10 (674 on wire, 674 captured) Arrival Time: Apr 23, 2003 15:28:20.284421000 Linux cooked capture Packet type: Sent by us (4) Link-layer address type: 1 Link-layer address length: 6 Protocol: IP (0x0800) Internet Protocol, Src Addr: dhcp164-devel (172.18.3.164), Dst Addr: shot.e-horizon.se (172.18.1.66) Version: 4 Header length: 20 bytes Protocol: UDP (0x11) Header checksum: 0xdb50 (correct) Source: dhcp164-devel (172.18.3.164) Destination: shot.e-horizon.se (172.18.1.66) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 7203 (7203) Source port: 5060 (5060) Destination port: 7203 (7203) Session Initiation Protocol Request line: MESSAGE sip:172.18.1.66:7203 SIP/2.0 Message Header Record-Route: <sip:x@172.18.3.164:5060;ftag=949cc72a86e24d5dcb2bf51338531b31-4809;lr> Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bK8984.7c66ebe.0 Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bKa858.b1ed4bd6.0 To: sip:x@e-horizon.se From: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se>;tag=949cc72a86e24d5dcb2bf51338531b31-4809 CSeq: 10 MESSAGE Call-ID: 38231eab-9474(a)172.18.3.164 Content-Length: 20 User-Agent: Sip EXpress router(0.8.11pre13 (i386/linux)) Content-Type: text/plain Contact: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se> Session Description Protocol Invalid line: bbbbbbbbbbbbbbbbbbbb Frame 11 (474 on wire, 474 captured) Arrival Time: Apr 23, 2003 15:28:20.285190000 Linux cooked capture Packet type: Unicast to us (0) Link-layer address type: 1 Link-layer address length: 6 Source: 00:05:5d:03:03:1e (D-Link_03:03:1e) Protocol: IP (0x0800) Internet Protocol, Src Addr: shot.e-horizon.se (172.18.1.66), Dst Addr: dhcp164-devel (172.18.3.164) Time to live: 127 Protocol: UDP (0x11) Header checksum: 0x765a (correct) Source: shot.e-horizon.se (172.18.1.66) Destination: dhcp164-devel (172.18.3.164) User Datagram Protocol, Src Port: 3173 (3173), Dst Port: 5060 (5060) Source port: 3173 (3173) Destination port: 5060 (5060) Length: 438 Checksum: 0x8c6c (correct) Session Initiation Protocol Status line: SIP/2.0 400 Bad Request Message Header Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bK8984.7c66ebe.0 Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bKa858.b1ed4bd6.0 From: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se>;tag=949cc72a86e24d5dcb2bf51338531b31-4809 To: sip:x@e-horizon.se;tag=199332f8-e402-4795-833a-c90969c979b0 Call-ID: 38231eab-9474(a)172.18.3.164 CSeq: 10 MESSAGE User-Agent: Windows RTC/1.0 Content-Length: 0 Frame 12 (409 on wire, 409 captured) Arrival Time: Apr 23, 2003 15:28:20.288277000 Linux cooked capture Packet type: Unicast to us (0) Link-layer address type: 772 Link-layer address length: 0 Source: <MISSING> Protocol: IP (0x0800) Internet Protocol, Src Addr: dhcp164-devel (172.18.3.164), Dst Addr: dhcp164-devel (172.18.3.164) Time to live: 64 Protocol: UDP (0x11) Header checksum: 0xd9f7 (correct) Source: dhcp164-devel (172.18.3.164) Destination: dhcp164-devel (172.18.3.164) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Length: 373 Checksum: 0x4b0e (correct) Session Initiation Protocol Status line: SIP/2.0 400 Bad Request Message Header Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bKa858.b1ed4bd6.0 From: <sip:b*storstark.e-horizon.se@jstorstark.e-horizon.se>;tag=949cc72a86e24d5dcb2bf51338531b31-4809 To: sip:x@e-horizon.se;tag=199332f8-e402-4795-833a-c90969c979b0 Call-ID: 38231eab-9474(a)172.18.3.164 CSeq: 10 MESSAGE User-Agent: Windows RTC/1.0 Content-Length: 0 . . . . Frame 19 (530 on wire, 530 captured) Frame Number: 19 Internet Protocol, Src Addr: dhcp164-devel (172.18.3.164), Dst Addr: dhcp164-devel (172.18.3.164) Version: 4 Header length: 20 bytes Time to live: 64 Protocol: UDP (0x11) Source: dhcp164-devel (172.18.3.164) Destination: dhcp164-devel (172.18.3.164) User Datagram Protocol, Src Port: 5060 (5060), Dst Port: 5060 (5060) Source port: 5060 (5060) Destination port: 5060 (5060) Session Initiation Protocol Request line: MESSAGE sip:x@e-horizon.se SIP/2.0 Message Header Via: SIP/2.0/UDP 172.18.3.164:5060;branch=z9hG4bKb649.52575b6.0 To: sip:x@e-horizon.se From: <sip:sip_to_jabber_gateway>;tag=949cc72a86e24d5dcb2bf51338531b31-dc30 CSeq: 10 MESSAGE Call-ID: 38231eac-9474(a)172.18.3.164 Content-Length: 85 User-Agent: Sip EXpress router(0.8.11pre13 (i386/linux)) Content-Type: text/plain Contact: <sip:sip_to_jabber_gateway> Session Description Protocol Invalid line: INFO: Your are now offline in Jabber network. Thank you for using SIP-Jabber gateway.

21 years, 2 months

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[Serusers] serweb & usrloc

by kelch

hi! is serweb intended to just send IM one-way? i configured everything and noticed that i can only send messages from a web-client (via serweb) to a sip softphone, but not the other way around. i noticed that this is due to the fact that when a user logs on to serweb, it does not put anything in the location table in ser db. so if a message was sent intended for that user, the sender receives 404 not found (due to failure in lookup()). what i am thinking as a workaround is to manipulate the php files in serweb to insert the user into the 'location' table via fifo, the method though crude might just work... my problem is, before i could try to do the workaround is whether serweb would accept the message if it indeed received one on port 5060? would it? or should the proxy send the message through another port like 80?

21 years, 2 months

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[Serusers] MOTOROLA T720i ��:229$

by 236553＠delphi.com

21 years, 2 months

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Re: [Serusers] Testing Concurrent Calls

by Jiri Kuthan

I'm not aware of an affordable stress tools which covers both media and signaling. -Jiri At 03:50 PM 4/24/2003, Alejandro Olchik wrote: >I would like to start multiple simultaneous calls >(let's say around 30), and keep calls active to >test concurrency. > >Any suggestion how can I do that without requiring >expensive software for that? > >I have enough ports to terminate the calls using >G729 but I don't know what to use for originate them. > >Regards, >Alejandro > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

21 years, 2 months

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[Serusers] how configure ser for pstn?

by Yang Xiang

Hi all, I put following config in the main routing-block for forwarding INVITE-msg to pstn-gatway: ... route{ ... if (!t_relay()) { sl_reply_error(); }; # forward message to PSTN Gateway if (uri=~"^sip:[0-9]*@"){ log(5, "Forward to pstn \n"); forward("10.20.0.6"); break; }; } However, SER doesn't do it and sends a reply "404 not found" back instead. Is it the correct position to put the function "forward()" there? By the way, ser also doesn't write log message to syslog. Following is my syslog.conf, is anything wrong in the syslog.conf? #ident "@(#)syslog.conf 1.5 98/12/14 SMI" /* SunOS 5.0 */ # # Copyright (c) 1991-1998 by Sun Microsystems, Inc. # All rights reserved. # # syslog configuration file. # # This file is processed by m4 so be careful to quote (`') names # that match m4 reserved words. Also, within ifdef's, arguments # containing commas must be quoted. # *.debug;*.notice;*.info;*.crit;*.alert /var/log/debug *.err;kern.notice;auth.notice /dev/sysmsg *.err;kern.debug;daemon.notice;mail.crit /var/adm/messages *.alert;kern.err;daemon.err operator *.alert root *.emerg * # if a non-loghost machine chooses to have authentication messages # sent to the loghost machine, un-comment out the following line: #auth.notice ifdef(`LOGHOST', /var/log/authlog, @loghost) mail.debug ifdef(`LOGHOST', /var/log/syslog, @loghost) # # non-loghost machines will use the following lines to cause "user" # log messages to be logged locally. # user.err /dev/sysmsg user.err /var/adm/messages user.alert `root, operator' user.emerg * auth.info /var/adm/messages local6.debug /var/adm/imapd.log auth.debug /var/adm/auth.log Thanks, Yang

21 years, 2 months

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[Serusers] Testing Concurrent Calls

by Alejandro Olchik

I would like to start multiple simultaneous calls (let's say around 30), and keep calls active to test concurrency. Any suggestion how can I do that without requiring expensive software for that? I have enough ports to terminate the calls using G729 but I don't know what to use for originate them. Regards, Alejandro

21 years, 2 months

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[Serusers] notify

by Kelvin Chua

how do you send notify? sl_send_reply("200","notify")? i don't think this is correct, i can't seem to find anything about it in the docs?

21 years, 2 months

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[Serusers] , about shared memory

by jimmy_huang

Hi , In Ser's Makefile.defs , originally , it define PKG_MALLOC , SHM_MEM , SHM_MMAP ,for shared memory, and I would like to marked it , cause i want to try let it not use shared memory (and modified ser.cfg let it just produce one main parent process ) I tried marked those three definitions , but it has error when compile , it seems some modules still need functions which only exist when those three definitions is defined, do those three definitions can't marked? or in ser , it can't work without using shared memory , can't mark those definition? Thanks , Jimmy

21 years, 2 months

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Re: [Serusers] Startup

by 'Jan Janak'

On 23-04 19:35, Dinesh wrote: > > Yes I do have "fork-yes" > > In cfg > > Should I change it to no? No. Send us your start-up script. And please reply to serusers(a)lists.iptel.org, not directly to me. Jan.

21 years, 2 months

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[Serusers] cisco 2600 as sip-pstn gateway

by Ng, Soo Sim

This is what you need 1 NM-2V + VIC-2BRI SSng

21 years, 2 months

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[Serusers] cisco 2600 as sip-pstn gateway

by Yang Xiang

Hi all, I want to use a cisco 2600 as sip2pstn gateway. In csico dokumentations and in postings of other people it seems that the cisco 2600 works with a PRI ISDN interface. Does anybody konw if it also works with BRI interface? I don't find a way to define "voice-port" on the router just with BRI interface. Has anybody here successfully tested this? Many thanks Yang

21 years, 2 months

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RE: [Serusers] cisco 2600 as sip-pstn gateway

by Dan Austin

The 'just a BRI' is the tricky part. You should need a NM-HDV (I think that's the part number), with a 2 port BRI VWIC. Dan -----Original Message----- From: Greg Fausak [mailto:greg@august.net] Sent: Wednesday, April 23, 2003 9:51 AM To: 'Yang Xiang'; serusers(a)lists.iptel.org Subject: RE: [Serusers] cisco 2600 as sip-pstn gateway Yes, it works with a BRI interface. I have experimented with a 3620 like that, the Same cards go in the 26xx series I believe. ---greg > -----Original Message----- > From: serusers-admin(a)lists.iptel.org > [mailto:serusers-admin@lists.iptel.org] On Behalf Of Yang Xiang > Sent: Wednesday, April 23, 2003 11:32 AM > To: serusers(a)lists.iptel.org > Subject: [Serusers] cisco 2600 as sip-pstn gateway > > > Hi all, > > I want to use a cisco 2600 as sip2pstn gateway. In csico > dokumentations and > in postings of other people it seems that the cisco 2600 > works with a PRI > ISDN interface. Does anybody konw if it also works with BRI > interface? I > don't find a way to define "voice-port" on the router just with BRI > interface. Has anybody here successfully tested this? > > Many thanks > > Yang > > > > _______________________________________________ > 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

21 years, 2 months

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[Serusers] fed up and need guidance

by Mat Harris

hi, i have been playing around with ser now for around a month and am getting nowhere. i have two sites running ser on redhat (7.3 and 8). the ser machines are each behind a redhat (7.3) firewall with port 5060(udp) forwarded to the ser machines. i cannot see presence of any users at the other site and vice versa. please can someone guide me through how they would set up such an operation. this really is a hand-holding job i'm afraid. very grateful for a quick response. cheers -- Mat Harris OpenGPG Public Key ID: C37D57D9 mat.harris(a)genestate.com www.genestate.com

21 years, 2 months

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Re: [Serusers] Groups

by 'Jan Janak'

Hello, On 23-04 19:28, Dinesh wrote: > Thanks for your comments but > > > 1. Could you confirm I would need to change the mysql db manually as > serctl restricts the group names it accepts Yes, edit serctl script and look for ACL_GROUPS variable, list the groups you want to use there. > 2. Could you give me a clue as to the script I would need to add. As I > cannot see how to check the called parties groups. is_user_in("To", "group") Jan.

21 years, 2 months

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[Serusers] Load Balancing

by Alejandro Olchik

I need some help to implement load balance when sending invites to an external sip proxy server. I would like to be able to balance load between to IPs (10.0.0.1 and 10.0.0.2) and use the backup IP when the primary one fails. Below is the routing code I have: if (!lookup("location")) { rewritehost("10.0.0.1"); if (!t_relay_to("10.0.0.1","5060")) { sl_send_reply("404", "Not Found"); }; break; }; How can I add this behaviour? Any suggestion? Alejandro

21 years, 2 months

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[Serusers] show active calls

by Bo

Is there any way to show active calls (dialed number, duration, originating IP...) with SER? Thanks.

21 years, 2 months

2
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[Serusers] RFC3261 Compliance

by Andrew Sze

Hello, I would like to know whether the current SER is RFC2543 or RFC3261 compliant. It is described to be a RFC3261 server in the webpage but is specified to be RFC2543 compliant in the product sheet. (I want to use SER to test with other RFC3261 devices.) Thank you. Andrew

21 years, 2 months

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Re: [Serusers] fed up and need guidance

by jaime.gill＠orange.co.uk

Hi Mat, Just saw your question posted in the mailing list. If I understand your scenario is like this: UA1 --- SER ---- NAT/FW --- Internet ---- NAT/FW ----- SER ---- UA2 If so, I have been working to get something like this working for a couple of months. I wrote a module that is supposed to act as a nathelper and as a FCP client. If you are still having problems to get this working, get back to me and I can provide you with guidance. Regards, Jaime Mat Harris <mat.harris(a)genestate.com> on 16/04/2003 14:40:35 To: serusers(a)lists.iptel.org cc: (bcc: Jaime GILL/EN/HTLUK) Subject: [Serusers] fed up and need guidance hi, i have been playing around with ser now for around a month and am getting nowhere. i have two sites running ser on redhat (7.3 and 8). the ser machines are each behind a redhat (7.3) firewall with port 5060(udp) forwarded to the ser machines. i cannot see presence of any users at the other site and vice versa. please can someone guide me through how they would set up such an operation. this really is a hand-holding job i'm afraid. very grateful for a quick response. cheers -- Mat Harris OpenGPG Public Key ID: C37D57D9 mat.harris(a)genestate.com www.genestate.com ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. *******************************************************************************

21 years, 2 months

1
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[Serusers] Illegal instruction

by Matthew McGillis

System Configuration: RedHat 8.0 Linux box Linux red.jenika.com 2.4.18-27.8.0 #1 Fri Mar 14 05:45:24 EST 2003 i586 i586 i386 GNU/Linux make-3.79.1-14 tar-1.13.25-8 bison-1.35-4 gcc-3.2-7 libgcc-3.2-7 flex-2.5.4a-26 Downloaded ser-0.8.10 source. make builds everything and install installed. However this is what I always get after executing it: $ /usr/local/sbin/ser Illegal instruction This is with debug: $ /usr/local/sbin/ser -dddd -E 0(25834) WARNING: hash function optimized for 1024 entries 0(25834) WARNING: use of 65536 entries may lead to unflat distribution 0(25834) qm_malloc_init: QM_OPTIMIZE=2048, /ROUNDTO=512 0(25834) qm_malloc_init: QM_HASH_SIZE=534, qm_block size=25656 0(25834) qm_malloc_init(0x80a5700, 1048576), start=0x80a5700 0(25834) qm_malloc_init: size= 1048576, init_overhead=25704 0(25834) qm_malloc_init: QM_OPTIMIZE=2048, /ROUNDTO=512 0(25834) qm_malloc_init: QM_HASH_SIZE=534, qm_block size=25656 0(25834) qm_malloc_init(0x4015d000, 33554432), start=0x4015d000 0(25834) qm_malloc_init: size= 33554432, init_overhead=25704 0(25834) qm_malloc(0x4015d000, 4) called from mem/shm_mem.c: shm_mem_init(275) 0(25834) qm_malloc(0x4015d000, 4) returns address 0x40163450 on 0 -th hit 0(25834) shm_mem_init: success 0(25834) qm_malloc(0x4015d000, 4) called from timer.c: init_timer(52) 0(25834) qm_malloc(0x4015d000, 4) returns address 0x40163484 on 0 -th hit 0(25834) DEBUG: register_fifo_cmd: new command (print) registered 0(25834) DEBUG: register_fifo_cmd: new command (uptime) registered 0(25834) DEBUG: register_fifo_cmd: new command (version) registered 0(25834) DEBUG: register_fifo_cmd: new command (which) registered 0(25834) DEBUG: register_fifo_cmd: new command (ps) registered Illegal instruction This is with strace: $ strace /usr/local/sbin/ser execve("/usr/local/sbin/ser", ["/usr/local/sbin/ser"], [/* 24 vars */]) = 0 uname({sys="Linux", node="red.jenika.com", ...}) = 0 brk(0) = 0x81a5760 open("/etc/ld.so.preload", O_RDONLY) = -1 ENOENT (No such file or directory) open("/etc/ld.so.cache", O_RDONLY) = 3 fstat64(3, {st_mode=S_IFREG|0644, st_size=44712, ...}) = 0 old_mmap(NULL, 44712, PROT_READ, MAP_PRIVATE, 3, 0) = 0x40013000 close(3) = 0 open("/lib/libdl.so.2", O_RDONLY) = 3 read(3, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\270\25"..., 512) = 512 fstat64(3, {st_mode=S_IFREG|0755, st_size=12165, ...}) = 0 old_mmap(NULL, 12116, PROT_READ|PROT_EXEC, MAP_PRIVATE, 3, 0) = 0x4001e000 old_mmap(0x40020000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, 0x1000) = 0x40020000 close(3) = 0 open("/lib/libresolv.so.2", O_RDONLY) = 3 read(3, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\230\'\0"..., 512) = 512 fstat64(3, {st_mode=S_IFREG|0755, st_size=69443, ...}) = 0 old_mmap(NULL, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_ANONYMOUS, -1, 0) = 0x40021000 old_mmap(NULL, 69444, PROT_READ|PROT_EXEC, MAP_PRIVATE, 3, 0) = 0x40022000 old_mmap(0x40030000, 4096, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, 0xe000) = 0x40030000 old_mmap(0x40031000, 8004, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x40031000 close(3) = 0 open("/lib/libc.so.6", O_RDONLY) = 3 read(3, "\177ELF\1\1\1\0\0\0\0\0\0\0\0\0\3\0\3\0\1\0\0\0\210U\1"..., 512) = 512 fstat64(3, {st_mode=S_IFREG|0755, st_size=1377691, ...}) = 0 old_mmap(NULL, 1219172, PROT_READ|PROT_EXEC, MAP_PRIVATE, 3, 0) = 0x40033000 old_mmap(0x40155000, 24576, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED, 3, 0x121000) = 0x40155000 old_mmap(0x4015b000, 6756, PROT_READ|PROT_WRITE, MAP_PRIVATE|MAP_FIXED|MAP_ANONYMOUS, -1, 0) = 0x4015b000 close(3) = 0 munmap(0x40013000, 44712) = 0 rt_sigaction(SIGINT, {0x80587b0, [INT], SA_RESTORER|SA_RESTART, 0x40059cf8}, {SIG_DFL}, 8) = 0 rt_sigaction(SIGPIPE, {0x80587b0, [PIPE], SA_RESTORER|SA_RESTART, 0x40059cf8}, {SIG_DFL}, 8) = 0 rt_sigaction(SIGUSR1, {0x80587b0, [USR1], SA_RESTORER|SA_RESTART, 0x40059cf8}, {SIG_DFL}, 8) = 0 rt_sigaction(SIGCHLD, {0x80587b0, [CHLD], SA_RESTORER|SA_RESTART, 0x40059cf8}, {SIG_DFL}, 8) = 0 rt_sigaction(SIGTERM, {0x80587b0, [TERM], SA_RESTORER|SA_RESTART, 0x40059cf8}, {SIG_DFL}, 8) = 0 rt_sigaction(SIGHUP, {0x80587b0, [HUP], SA_RESTORER|SA_RESTART, 0x40059cf8}, {SIG_DFL}, 8) = 0 rt_sigaction(SIGUSR2, {0x80587b0, [USR2], SA_RESTORER|SA_RESTART, 0x40059cf8}, {SIG_DFL}, 8) = 0 brk(0) = 0x81a5760 brk(0x81a6760) = 0x81a6760 brk(0) = 0x81a6760 brk(0x81a7000) = 0x81a7000 open("/usr/local/etc/ser/ser.cfg", O_RDONLY) = 3 open("/dev/zero", O_RDWR) = 4 old_mmap(NULL, 33554432, PROT_READ|PROT_WRITE, MAP_SHARED, 4, 0) = 0x4015d000 close(4) = 0 brk(0) = 0x81a7000 brk(0x81aa000) = 0x81aa000 ioctl(3, SNDCTL_TMR_TIMEBASE, 0xbfffd2e0) = -1 ENOTTY (Inappropriate ioctl for device) --- SIGILL (Illegal instruction) --- +++ killed by SIGILL +++ This is the stack trace from gdb: $ gdb /usr/local/sbin/ser GNU gdb Red Hat Linux (5.2.1-4) Copyright 2002 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "i386-redhat-linux"... (gdb) run Starting program: /usr/local/sbin/ser Program received signal SIGILL, Illegal instruction. 0x0807204b in yy_get_next_buffer () (gdb) bt #0 0x0807204b in yy_get_next_buffer () #1 0x08071cf8 in yylex () #2 0x080750ce in yyparse () #3 0x0805634e in main () #4 0x400484ed in __libc_start_main () from /lib/libc.so.6 Any ideas or thoughts would be appreciated please respond to my e-mail address I'm not a member of this mail list. Thanks! Matthew McGillis

21 years, 2 months

2
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[Serusers] SIMPLE Features of ser

by Steve Lewis

Has anyone tried out the SIMPLE features of ser? Based on what I have read it appears to support contacts and presence. If so, which clients has it been tested with? Regards, Steve

21 years, 2 months

2
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RE: [Serusers] pstn connectivity

by LeeCC, Kenny

instead of t_relay_to("xxx.xxx.xxx.xxx","5060"); try rewritehostport("xxx.xxx.xxx.xxx:5060"); > -----Original Message----- > From: budi wibowo [mailto:bu1d@yahoo.com] > Sent: Friday, April 18, 2003 09:54 > To: serusers(a)lists.iptel.org > Subject: [Serusers] pstn connectivity > > > hi i'm new guy in this list > and i have a litle problem with connectivity to psnt > thru cisco as5300 > below is my ser.cfg, yyy.yyy.yyy.yyy is ip of ser and > xx.xxx.xxx.xxx is ip of the cisco gw > > if (uri=~"^sip:[0-9]*@yyy.yyy.yyy.yyy"){ > log("Forward to pstn \n"); > t_relay_to("xxx.xxx.xxx.xxx","5060"); > break; > }; > > > > and i have this on my cisco > ! > dial-peer voice 4 voip > incoming called number . > destination-pattern 2525T > session protocol sipv2 > session target sip-server > codec g723r63 > ! > ! > sip-ua > calling-info sip-to-pstn number set 113950029 > sip-server ipv4:yyy.yyy.yy.yyyy > ! > > my pproblem is.. i cant if i dial to 252544123 from my > softphone , it's not work > but when i dial with format 252544123(a)xxx.xxx.xxx.xxx > it always work > anyone have clue for my problem? > thx > > budi > > > __________________________________________________ > Do you Yahoo!? > The New Yahoo! Search - Faster. Easier. Bingo > http://search.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

21 years, 2 months

5
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[Serusers] build static modules , undefined reference to 'usrloc_exports' ?

by jimmy_huang

hello... I want to build a small and workable ser (let user can sending messages , registering , talking via ser...) , and build needed modules to static modules , what modules should I build in with static modules? I tried built tm,sl,rr,maxfwd,usrloc these modules to static modules , but it has some error...in compiling sr_module.o only when build usrloc modules to static modules, does "usrloc" modules can't build to static modules? or i have missing something?? error msg as follow... sr_module.o (.text+0x5a9):In function 'register_builtin_modules': :undefined reference to 'usrloc_exports' My environment is in RedHat 9.0 Jimmy

21 years, 2 months

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RE: [Serusers] pstn connectivity

by Dan Austin

OK. It sounds like the Softphone is not putting the realm/domain in the URI. Which softphone are you using? -----Original Message----- From: budi wibowo [mailto:bu1d@yahoo.com] Sent: Saturday, April 19, 2003 7:29 AM To: Dan Austin Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] pstn connectivity yes i have resistered my softphone, if i connect using the wrong username or the wrong password i got message like not authorized how do i check if my softphoen is registered or not if i dial to 1136523456(a)111.111.111.111 everything is normal tia budi --- Dan Austin <Dan_Austin(a)Phoenix.com> wrote: > Is the softphone configured to register with your > SER install? > > I've verified the setting in the how-to with MSN > messenger 4.6 > and a Cisco 7940 converted to use SIP. > > If you have to add the ip address of the gw to the > request, it > would seem like your phone might not be registered. > What happens > if you send the call to 1136523456(a)111.111.111.111? > > Dan > > -----Original Message----- > From: budi wibowo [mailto:bu1d@yahoo.com] > Sent: Friday, April 18, 2003 7:53 PM > To: serusers(a)lists.iptel.org > Subject: Re: [Serusers] pstn connectivity > > > - the request sent by the softphone > - to check the request on server ..how i do that > - i do debug ccsip all on my gateway and i saw > something uncomplete for the request , like the > translation rule not work properly > but i include the ip of cisco gw eerything is normal > any clue? or any suggestion which softphone i should > use? > > > boedz > > > > > --- Nils Ohlmeier <nils(a)iptel.org> wrote: > > Sounds like you softphone do not know where to > send > > a request if you omit the > > host part. Is the request sent by the softphone if > > you omit the host part? Is > > the request received at the server? Do the server > > deliver the request if it > > received it? and so on and on and on... > > > > Greetings > > Nils > > > > On Saturday 19 April 2003 04:31, budi wibowo > wrote: > > > dear all i repeat my problem > > > - pc running ser with ip 111.111.111.111 > > > - cisco 5350 with dial-peer to sip-server with > ip > > > 222.222.222.222 > > > - remote pc running sip softphone > > > > > > > > > i have read austin how to.. regarding how to > > connect > > > to cisco gw > > > but it's not work > > > example ..i want to dial to singapore with > pattern > > 113 > > > if i dial from my softphone 1136523456 it always > > > failed, but if dial using > > 1136523456(a)222.222.222.222 > > > it always work . > > > is there anything missing on my config? > > > > > > > > > rgds > > > > > > budi > > > __________________________________________________ > Do you Yahoo!? > The New Yahoo! Search - Faster. Easier. Bingo > http://search.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers __________________________________________________ Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo http://search.yahoo.com

21 years, 2 months

3
2
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RE: [Serusers] pstn connectivity

by budi wibowo

yes i have resistered my softphone, if i connect using the wrong username or the wrong password i got message like not authorized how do i check if my softphoen is registered or not if i dial to 1136523456(a)111.111.111.111 everything is normal tia budi --- Dan Austin <Dan_Austin(a)Phoenix.com> wrote: > Is the softphone configured to register with your > SER install? > > I've verified the setting in the how-to with MSN > messenger 4.6 > and a Cisco 7940 converted to use SIP. > > If you have to add the ip address of the gw to the > request, it > would seem like your phone might not be registered. > What happens > if you send the call to 1136523456(a)111.111.111.111? > > Dan > > -----Original Message----- > From: budi wibowo [mailto:bu1d@yahoo.com] > Sent: Friday, April 18, 2003 7:53 PM > To: serusers(a)lists.iptel.org > Subject: Re: [Serusers] pstn connectivity > > > - the request sent by the softphone > - to check the request on server ..how i do that > - i do debug ccsip all on my gateway and i saw > something uncomplete for the request , like the > translation rule not work properly > but i include the ip of cisco gw eerything is normal > any clue? or any suggestion which softphone i should > use? > > > boedz > > > > > --- Nils Ohlmeier <nils(a)iptel.org> wrote: > > Sounds like you softphone do not know where to > send > > a request if you omit the > > host part. Is the request sent by the softphone if > > you omit the host part? Is > > the request received at the server? Do the server > > deliver the request if it > > received it? and so on and on and on... > > > > Greetings > > Nils > > > > On Saturday 19 April 2003 04:31, budi wibowo > wrote: > > > dear all i repeat my problem > > > - pc running ser with ip 111.111.111.111 > > > - cisco 5350 with dial-peer to sip-server with > ip > > > 222.222.222.222 > > > - remote pc running sip softphone > > > > > > > > > i have read austin how to.. regarding how to > > connect > > > to cisco gw > > > but it's not work > > > example ..i want to dial to singapore with > pattern > > 113 > > > if i dial from my softphone 1136523456 it always > > > failed, but if dial using > > 1136523456(a)222.222.222.222 > > > it always work . > > > is there anything missing on my config? > > > > > > > > > rgds > > > > > > budi > > > __________________________________________________ > Do you Yahoo!? > The New Yahoo! Search - Faster. Easier. Bingo > http://search.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers __________________________________________________ Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo http://search.yahoo.com

21 years, 2 months

1
0
0 0

[Serusers] Having difficulty running Serweb

by trung

Hi there, I have been reading Austin-how-to on the serweb and the forum, I manage to get the website up and running. But when I log into it as admin. The screen doesn't look like its getting me anywhere. It stay the same and ask to enter username and password again. If anyone have this serweb running successful on Redhat 8 or give me hint to get this running, it will be an honor. Thanks in Advance, Trung

21 years, 2 months

2
1
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[Serusers] error in rr.so

by Dopf Walter

hi all, in the new ver. 0.8.11 i changed additional to the news: rewriteFromRoute has been replaced with strict_route() and: 0(26945) loading module /usr/lib/ser/modules/rr.so but: 0(26945) find_export: <strict_route> not found 0(26945) find_export: <strict_route> not found 0(26945) parse error (75,15-16): unknown command, missing loadmodule? what's missing ? best wdo

21 years, 2 months

4
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[Serusers] failure in auth.so in ver 0.8.11

by Dopf Walter

hi all, i don't know what's going wrong here. i made an uprade to ver 0.8.11, since that time i get allways the same failure: 0(26268) set_mod_param_regex: auth matches module auth 0(26268) set_mod_param_regex: parameter <calculate_ha1> not found in module <auth> and 0(26268) set_mod_param_regex: auth matches module auth 0(26268) set_mod_param_regex: parameter <password_column> not found in module <auth> all commands former been handled by he module auth.so cause an error any ideas best wdo

21 years, 2 months

2
1
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[Serusers] pstn connectivity

by budi wibowo

hi i'm new guy in this list and i have a litle problem with connectivity to psnt thru cisco as5300 below is my ser.cfg, yyy.yyy.yyy.yyy is ip of ser and xx.xxx.xxx.xxx is ip of the cisco gw if (uri=~"^sip:[0-9]*@yyy.yyy.yyy.yyy"){ log("Forward to pstn \n"); t_relay_to("xxx.xxx.xxx.xxx","5060"); break; }; and i have this on my cisco ! dial-peer voice 4 voip incoming called number . destination-pattern 2525T session protocol sipv2 session target sip-server codec g723r63 ! ! sip-ua calling-info sip-to-pstn number set 113950029 sip-server ipv4:yyy.yyy.yy.yyyy ! my pproblem is.. i cant if i dial to 252544123 from my softphone , it's not work but when i dial with format 252544123(a)xxx.xxx.xxx.xxx it always work anyone have clue for my problem? thx budi __________________________________________________ Do you Yahoo!? The New Yahoo! Search - Faster. Easier. Bingo http://search.yahoo.com

21 years, 2 months

1
0
0 0

[Serusers] Grandstream Budgettel 100 problems

by Steve Lewis

Hi All, Has anyone got a Grandstream Budgettel SIP phone to work successfully with 8.10 version of SER. When Attempting to register I get an error with what appears to be the Allow header. Actual header value is from phone is (from ethereal capture is): Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, OPTIONS Log Entry Apr 17 10:39:52 mercedes /usr/local/sbin/ser[31505]: ERROR: get_hdr_field: bad body for <Allow>(1048576) Apr 17 10:39:52 mercedes /usr/local/sbin/ser[31505]: ERROR: bad header field Apr 17 10:39:52 mercedes /usr/local/sbin/ser[31505]: ERROR: build_res_buf_from_sip_req: alas, parse_headers failed Any help would be appreciated. Steve

21 years, 2 months

2
6
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[Serusers] refer

by Joe_chen

hi, i want to know whether ser-0.8.10 supports the REFER method? If it does not support,when can you release the new version of ser? thank you.

21 years, 2 months

2
1
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[Serusers] ENUM support

by Steve Lewis

I'm interesting in experimenting with ENUM in conjunction with Ser and was wondering if there is support for enum within ser. Additionally, if anyone has experience and/or information on setting up a dns server for use with ENUM that would be appreciated. Steve Lewis

21 years, 2 months

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