[SR-Users] Add custom parameter to To: header, and then recover it

2014-09-05 Thread Alex Villací­s Lasso
What functions exist (if any), to add custom parameters to the From: and To: headers? ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-user

Re: [SR-Users] Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)

2014-09-02 Thread Alex Villací­s Lasso
record routing log message about missing socket. Otherwise, you should ignore it if the sip routing is ok. Cheers, Daniel On 01/09/14 18:55, Alex Villací­s Lasso wrote: [...] Maybe I should explain my setup better. The test setup I want to run is, in a way, twice natted. The asterisk instance runs

Re: [SR-Users] Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)

2014-09-01 Thread Alex Villací­s Lasso
El 01/09/14 10:50, Alex Villací­s Lasso escribió: El 01/09/14 05:15, Daniel-Constantin Mierla escribió: On 29/08/14 23:58, Andres wrote: On 8/29/14, 1:42 PM, Alex Villací­s Lasso wrote: Please consider the following SIP packet exchange, as seen by a tcpdump running on 201.234.196.170. Here

Re: [SR-Users] Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)

2014-09-01 Thread Alex Villací­s Lasso
El 01/09/14 05:15, Daniel-Constantin Mierla escribió: On 29/08/14 23:58, Andres wrote: On 8/29/14, 1:42 PM, Alex Villací­s Lasso wrote: Please consider the following SIP packet exchange, as seen by a tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a call to 201.234.196.170

Re: [SR-Users] Kamailio as Websocket bridge to Asterisk, and Asterisk-sent OPTIONS

2014-08-29 Thread Alex Villací­s Lasso
El 29/08/14 14:44, Paul Belanger escribió: On Fri, Aug 29, 2014 at 11:55 AM, Alex Villací­s Lasso wrote: El 28/08/14 19:09, Paul Belanger escribió: On Thu, Aug 28, 2014 at 7:18 PM, Alex Villací­s Lasso wrote: As a continuation of my project, I am trying to set up Kamailio as a Websocket

[SR-Users] Help debugging a missing ACK (is Asterisk covering up a mistake in my Kamailio config?)

2014-08-29 Thread Alex Villací­s Lasso
Please consider the following SIP packet exchange, as seen by a tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a call to 201.234.196.170: IP 198.58.101.75.5060 > 201.234.196.170.5060 INVITE sip:*43@201.234.196.170:5060 SIP/2.0 Via: SIP/2.0/UDP 198.58.101.75:5060;branch=z9hG4bK7

Re: [SR-Users] Kamailio as Websocket bridge to Asterisk, and Asterisk-sent OPTIONS

2014-08-29 Thread Alex Villací­s Lasso
El 28/08/14 19:09, Paul Belanger escribió: On Thu, Aug 28, 2014 at 7:18 PM, Alex Villací­s Lasso wrote: As a continuation of my project, I am trying to set up Kamailio as a Websocket bridge to Asterisk. The asterisk instance is running as localhost, with its own websocket support disabled, but

[SR-Users] Kamailio as Websocket bridge to Asterisk, and Asterisk-sent OPTIONS

2014-08-28 Thread Alex Villací­s Lasso
As a continuation of my project, I am trying to set up Kamailio as a Websocket bridge to Asterisk. The asterisk instance is running as localhost, with its own websocket support disabled, but otherwise has accounts with all of the avfp and dtls settings for websockets. Additionally, I have removed

Re: [SR-Users] How do I translate rtpproxy bridge mode config to mediaproxy-ng/rtpengine?

2014-08-26 Thread Alex Villací­s Lasso
El 26/08/14 12:02, Alex Villací­s Lasso escribió: El 25/08/14 18:28, Alex Balashov escribió: On 08/25/2014 07:25 PM, Alex Villací­s Lasso wrote: However, I do not find an equivalent to bridge mode in the rtpengine command-line parameters. Bridging mode of this type is not supported by

Re: [SR-Users] How do I translate rtpproxy bridge mode config to mediaproxy-ng/rtpengine?

2014-08-26 Thread Alex Villací­s Lasso
El 25/08/14 18:28, Alex Balashov escribió: On 08/25/2014 07:25 PM, Alex Villací­s Lasso wrote: However, I do not find an equivalent to bridge mode in the rtpengine command-line parameters. Bridging mode of this type is not supported by rtpengine. If this is true, then mediaproxy-ng

[SR-Users] How do I translate rtpproxy bridge mode config to mediaproxy-ng/rtpengine?

2014-08-25 Thread Alex Villací­s Lasso
I have a rtpproxy configuration that spawns several rtpproxy instances, using bridge mode. An example is shown below: /usr/bin/rtpproxy -p /var/run/rtpproxy.pid-7723 -u rtpproxy -s udp:127.0.0.1 7723 192.168.2.18/127.0.0.1 -m 1 -M 2 Here, rtpproxy bridges between 192.168.2.18 and 127.0

Re: [SR-Users] subst_hf does unexpected append instead of a replace (WAS: How do I make subst_hf() use variables in search (not replacement)?)

2014-08-13 Thread Alex Villací­s Lasso
El 13/08/14 03:12, Daniel-Constantin Mierla escribió: Would be nicer not to forward all headers inline, that results in a message easy to read (especially on mobile devices) and therefore faster to answer. You use subst over contact and set contact allias, which both append values there, deleti

[SR-Users] subst_hf does unexpected append instead of a replace (WAS: How do I make subst_hf() use variables in search (not replacement)?)

2014-08-12 Thread Alex Villací­s Lasso
014 14:02:09 -0500 From: Alex Villací­s Lasso User-Agent: Mozilla/5.0 (X11; Linux x86_64; rv:24.0) Gecko/20100101 Thunderbird/24.7.0 MIME-Version: 1.0 To: sr-users@lists.sip-router.org References: <53e54348.9090...@palosanto.com> <53e8712a.2050...@gmail.com> In-Reply-To:

Re: [SR-Users] How do I make subst_hf() use variables in search (not replacement)?

2014-08-11 Thread Alex Villací­s Lasso
stdef "/ASTERISKPORT/5080/" asterisk.bindip = "ASTERISKIP" ... asterisk.bindport = ASTERISKPORT ... subst_hf("Contact", "/ASTERISKIP:ASTERISKPORT/$td/", "a"); Cheers, Daniel On 08/08/14 23:38, Alex Villací­s Lasso wrote: Consider the fol

[SR-Users] How do I make subst_hf() use variables in search (not replacement)?

2014-08-08 Thread Alex Villací­s Lasso
Consider the following snippet: if (is_present_hf("Contact")) { xlog("L_ALERT", "= reply to SUBSCRIBE has Contact: $ct\n"); xlog("L_ALERT", "= want to replace with $td\n"); xlog("L_ALERT", "= regexp to use is /$sel(cfg_get.asterisk.bindip):$sel(cfg_get.asteris

Re: [SR-Users] kamailio routes packets with invalid From/To headers with uac.restore_mode=auto when incoming packet does not use exact same replaced From/To header

2014-08-01 Thread Alex Villací­s Lasso
El 01/08/14 01:24, Daniel-Constantin Mierla escribió: The uac from/to replacement relies that parties keep the same from/to headers content. The mechanism to replace A with B is to combine both and get the key X which is added in the record-route as parameter. Then practically from A and X res

[SR-Users] (SOLVED) Re: Unable to SUBSCRIBE for presence using sip.js through WSS

2014-07-31 Thread Alex Villací­s Lasso
El 31/07/14 11:11, Alex Villací­s Lasso escribió: El 30/07/14 17:33, Alex Villací­s Lasso escribió: My kamailio.cfg configuration file is attached. I am having trouble using SIP.js (http://sipjs.com/) to handle a SUBSCRIBE for presence information. With Jitsi clients (using plain UDP

Re: [SR-Users] Unable to SUBSCRIBE for presence using sip.js through WSS

2014-07-31 Thread Alex Villací­s Lasso
El 30/07/14 17:33, Alex Villací­s Lasso escribió: My kamailio.cfg configuration file is attached. I am having trouble using SIP.js (http://sipjs.com/) to handle a SUBSCRIBE for presence information. With Jitsi clients (using plain UDP), presence seems to work correctly. However, when using

[SR-Users] Unable to SUBSCRIBE for presence using sip.js through WSS

2014-07-30 Thread Alex Villací­s Lasso
My kamailio.cfg configuration file is attached. I am having trouble using SIP.js (http://sipjs.com/) to handle a SUBSCRIBE for presence information. With Jitsi clients (using plain UDP), presence seems to work correctly. However, when using SIP.js via a websocket, Kamailio is unable to send the N

[SR-Users] kamailio routes packets with invalid From/To headers with uac.restore_mode=auto when incoming packet does not use exact same replaced From/To header

2014-07-30 Thread Alex Villací­s Lasso
I am currently handling a system that runs kamailio and asterisk in the same machine. The kamailio instances are being used to emulate multiple SIP domains, by means of From/To mangling of incoming packets, which are then routed to Asterisk. The attached kamailio.cfg does this work. There is an

[SR-Users] SIGUSR1 for memory status not working as documented - only one process reports back

2014-07-30 Thread Alex Villací­s Lasso
I am trying to track down a memory leak that was triggered by a patch I wrote for my local copy of kamailio 4.1.4 . For this, I am following the documentation at http://www.kamailio.org/dokuwiki/doku.php/troubleshooting:memory . This page claims that once memlog is set in the configuration file, a

Re: [SR-Users] Many instances of "new URI shorter than old URI" in log

2014-07-15 Thread Alex Villací­s Lasso
El 02/07/14 15:02, Alex Villací­s Lasso escribió: El 02/07/14 10:51, Alex Villací­s Lasso escribió: El 30/06/14 04:24, Daniel-Constantin Mierla escribió: Hello, iirc, this is a check when trying to convert from base64 to old uri, but I didn't have time to dig properly inside the source

Re: [SR-Users] Many instances of "new URI shorter than old URI" in log

2014-07-02 Thread Alex Villací­s Lasso
El 02/07/14 10:51, Alex Villací­s Lasso escribió: El 30/06/14 04:24, Daniel-Constantin Mierla escribió: Hello, iirc, this is a check when trying to convert from base64 to old uri, but I didn't have time to dig properly inside the sources. I enhanced the log message to print the values o

Re: [SR-Users] Making RLS presence work with Blink and Kamailio 4.1.4

2014-07-02 Thread Alex Villací­s Lasso
El 01/07/14 14:44, Alex Villací­s Lasso escribió: El 26/06/14 18:39, Alex Villací­s Lasso escribió: I am having trouble making all of the supposed features of Blink work with Kamailio 4.1.4. My kamailio.cfg file is attached. Specifically, what I am having trouble is with presence (the way Blink

Re: [SR-Users] Many instances of "new URI shorter than old URI" in log

2014-07-02 Thread Alex Villací­s Lasso
El 30/06/14 04:24, Daniel-Constantin Mierla escribió: Hello, iirc, this is a check when trying to convert from base64 to old uri, but I didn't have time to dig properly inside the sources. I enhanced the log message to print the values of the uris, via commit: - http://git.sip-router.org/cgi

Re: [SR-Users] Making RLS presence work with Blink and Kamailio 4.1.4

2014-07-01 Thread Alex Villací­s Lasso
El 26/06/14 18:39, Alex Villací­s Lasso escribió: I am having trouble making all of the supposed features of Blink work with Kamailio 4.1.4. My kamailio.cfg file is attached. Specifically, what I am having trouble is with presence (the way Blink wants to implement it), and MSRP. Ordinary voice

Re: [SR-Users] Making RLS presence work with Blink and Kamailio 4.1.4

2014-06-30 Thread Alex Villací­s Lasso
El 28/06/14 00:31, Juha Heinanen escribió: there is at least one show stopper to make blink rls to work with any standards compliant implementation. this is tracker item that i opened year ago. Do you have an URL for this tracker item? -- juha i tried to test address book stuff with blink, bu

[SR-Users] Many instances of "new URI shorter than old URI" in log

2014-06-27 Thread Alex Villací­s Lasso
Jun 27 15:28:42 elx /usr/sbin/kamailio[25189]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:44 elx /usr/sbin/kamailio[25187]: ERROR: uac [replace.c:590]: restore_uri(): new URI shorter than old URI Jun 27 15:28:48 elx /usr/sbin/kamailio[25188]: ERROR: uac

[SR-Users] Making RLS presence work with Blink and Kamailio 4.1.4

2014-06-26 Thread Alex Villací­s Lasso
I am having trouble making all of the supposed features of Blink work with Kamailio 4.1.4. My kamailio.cfg file is attached. Specifically, what I am having trouble is with presence (the way Blink wants to implement it), and MSRP. Ordinary voice calls work correctly. With presence, I have managed

[SR-Users] How to route on whether INVITE is for MSRP

2014-06-23 Thread Alex Villací­s Lasso
So far, I have a kamailio.cfg that routes INVITEs between Asterisk and the external networks. Now I want to start adding MSRP support. However, with my current configuration, the test client (Blink) sends an INVITE with a SDP payload that specifies msrps media. This gets routed to Asterisk, and it

[SR-Users] Warning message on SIP websocket

2014-06-23 Thread Alex Villací­s Lasso
Just a quick question. What is supposed to be wrong with the below message? Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: [parser/parse_fline.c:243]: parse_first_line(): parse_first_line: bad message (offset: 22) Jun 23 12:47:16 elx /usr/sbin/kamailio[20891]: ERROR: [parser/msg_parser

Re: [SR-Users] SOLVED (PATCH): Re: Kamailio accepts and stores XCAP XML document from jitsi but rejects it on retrieval

2014-06-11 Thread Alex Villací­s Lasso
El 11/06/14 08:47, Daniel-Constantin Mierla escribió: Hello, can you put the patch on the tracker to review it -- in this way should not be forgotten. - http://sip-router.org/tracker/ I looked a bit, but has some new functions and touches quite a lot the existing parts, so I need more time t

Re: [SR-Users] SOLVED (PATCH): Re: Kamailio accepts and stores XCAP XML document from jitsi but rejects it on retrieval

2014-06-10 Thread Alex Villací­s Lasso
El 03/06/14 14:09, Alex Villací­s Lasso escribió: El 02/06/14 12:31, Alex Villací­s Lasso escribió: El 31/05/14 01:30, Daniel-Constantin Mierla escribió: On 30/05/14 22:02, Alex Villací­s Lasso wrote: [...] The xml document retrieval is failing again. Now it is because retrieval is

[SR-Users] SOLVED (PATCH): Re: Kamailio accepts and stores XCAP XML document from jitsi but rejects it on retrieval

2014-06-03 Thread Alex Villací­s Lasso
El 02/06/14 12:31, Alex Villací­s Lasso escribió: El 31/05/14 01:30, Daniel-Constantin Mierla escribió: On 30/05/14 22:02, Alex Villací­s Lasso wrote: [...] The xml document retrieval is failing again. Now it is because retrieval is truncating the MEDIUMTEXT field at 1024 characters: Adjust

Re: [SR-Users] Kamailio accepts and stores XCAP XML document from jitsi but rejects it on retrieval

2014-06-02 Thread Alex Villací­s Lasso
El 31/05/14 01:30, Daniel-Constantin Mierla escribió: On 30/05/14 22:02, Alex Villací­s Lasso wrote: [...] The xml document retrieval is failing again. Now it is because retrieval is truncating the MEDIUMTEXT field at 1024 characters: Adjust the value of parameter: http://kamailio.org/docs

Re: [SR-Users] Kamailio accepts and stores XCAP XML document from jitsi but rejects it on retrieval

2014-05-30 Thread Alex Villací­s Lasso
El 30/05/14 12:42, Alex Villací­s Lasso escribió: El 30/05/14 11:02, Alex Villací­s Lasso escribió: El 29/05/14 12:15, Alex Villací­s Lasso escribió: El 29/05/14 07:51, Daniel-Constantin Mierla escribió: Hello, the error message is printed in case of an invalid xml document. Can you edit

Re: [SR-Users] Kamailio accepts and stores XCAP XML document from jitsi but rejects it on retrieval

2014-05-30 Thread Alex Villací­s Lasso
El 30/05/14 11:02, Alex Villací­s Lasso escribió: El 29/05/14 12:15, Alex Villací­s Lasso escribió: El 29/05/14 07:51, Daniel-Constantin Mierla escribió: Hello, the error message is printed in case of an invalid xml document. Can you edit modules/xcap_server/xcap_misc.c and add inside

Re: [SR-Users] Kamailio accepts and stores XCAP XML document from jitsi but rejects it on retrieval

2014-05-30 Thread Alex Villací­s Lasso
El 29/05/14 12:15, Alex Villací­s Lasso escribió: El 29/05/14 07:51, Daniel-Constantin Mierla escribió: Hello, the error message is printed in case of an invalid xml document. Can you edit modules/xcap_server/xcap_misc.c and add inside function: int xcaps_check_doc_validity(str *doc) the

Re: [SR-Users] Kamailio accepts and stores XCAP XML document from jitsi but rejects it on retrieval

2014-05-29 Thread Alex Villací­s Lasso
El 29/05/14 07:51, Daniel-Constantin Mierla escribió: Hello, the error message is printed in case of an invalid xml document. Can you edit modules/xcap_server/xcap_misc.c and add inside function: int xcaps_check_doc_validity(str *doc) the log: LM_ERR("xmld document is: [[%.*s]]\n", doc->len

[SR-Users] Kamailio accepts and stores XCAP XML document from jitsi but rejects it on retrieval

2014-05-28 Thread Alex Villací­s Lasso
I am trying to set up XCAP support for SIP SIMPLE, and using jitsi as the sample client. The kamailio database was previously configured using ODBC and kamailio-4.1.3 and was apparently working correctly. I was following the guide at http://nil.uniza.sk/instant-messaging/simple/configuring-xcap-su

Re: [SR-Users] Possible causes of calls being terminated (ACK packet never received)

2014-05-27 Thread Alex Villací­s Lasso
El 26/05/14 11:05, Alex Villací­s Lasso escribió: My current working theory is that the Record-Route headers are incorrect in the original trace. The trace, as captured in the firewall, looks like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 38.126.208.41:5060;rport=5060;branch

Re: [SR-Users] Possible causes of calls being terminated (ACK packet never received)

2014-05-26 Thread Alex Villací­s Lasso
My current working theory is that the Record-Route headers are incorrect in the original trace. The trace, as captured in the firewall, looks like this: SIP/2.0 200 OK Via: SIP/2.0/UDP 38.126.208.41:5060;rport=5060;branch=b18bbcc394bb5dae43946b12f5d5fe0e Record-Route: Record-Route: The f

Re: [SR-Users] Possible causes of calls being terminated (ACK packet never received)

2014-05-23 Thread Alex Villací­s Lasso
El 21/05/14 11:41, Alex Villací­s Lasso escribió: El 21/05/14 10:28, Alex Villací­s Lasso escribió: El 21/05/14 00:52, Juha Heinanen escribió: Alex Villací­s Lasso writes: I am trying to explain the situation to our carrier, but I want to rule out possible misconfigurations on our side. Are

Re: [SR-Users] Possible causes of calls being terminated (ACK packet never received)

2014-05-21 Thread Alex Villací­s Lasso
El 21/05/14 10:28, Alex Villací­s Lasso escribió: El 21/05/14 00:52, Juha Heinanen escribió: Alex Villací­s Lasso writes: I am trying to explain the situation to our carrier, but I want to rule out possible misconfigurations on our side. Are there common misconfigurations that produce the

Re: [SR-Users] Possible causes of calls being terminated (ACK packet never received)

2014-05-21 Thread Alex Villací­s Lasso
El 21/05/14 00:52, Juha Heinanen escribió: Alex Villací­s Lasso writes: I am trying to explain the situation to our carrier, but I want to rule out possible misconfigurations on our side. Are there common misconfigurations that produce the symptoms described here? Are there any issues evident

[SR-Users] Possible causes of calls being terminated (ACK packet never received)

2014-05-20 Thread Alex Villací­s Lasso
I am trying to diagnose a SIP issue between our carrier and our network. The carrier has a CARRIER_IP and a different CARRIER_MEDIA_IP, and it submits an INVITE packet to MY_PUBLIC_IP using MY_DID. The firewall at our public IP (where the attached traffic sample was taken) redirects it to a partic

[SR-Users] Help diagnosing non-response from incoming call

2014-05-09 Thread Alex Villací­s Lasso
I have this setup for kamailio + asterisk, in which kamailio is supposed to listen on all ethernet interfaces on UDP port 5060, and will forward traffic from/to asterisk running on the same machine, and listening on localhost, udp port 5080. The scenario for the problematic call is somewhat like t

[SR-Users] Multiple port configuration

2014-05-09 Thread Alex Villací­s Lasso
How do I configure kamailio to listen to multiple ports, on all known interfaces? I have tried "listen=udp:*:5060", "listen=udp:*:5062" on separate lines, but I get the following errors: Not starting : invalid configuration file! 0(7806) : [cfg.y:3411]: yyerror_at(): parse error in config fi

[SR-Users] SOLVED: Re: Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

2014-05-02 Thread Alex Villací­s Lasso
El 02/05/14 10:40, Alex Villací­s Lasso escribió: El 24/04/14 19:09, Alex Villací­s Lasso escribió: I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration (attached) was based on the reference at http://kb.asipto.com

Re: [SR-Users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

2014-05-02 Thread Alex Villací­s Lasso
El 24/04/14 19:09, Alex Villací­s Lasso escribió: I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration (attached) was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0

[SR-Users] Proper way to make Asterisk recognize SIP trunk of incoming INVITE when IP is not available

2014-04-24 Thread Alex Villací­s Lasso
I am currently preparing a kamailio-asterisk combination. The asterisk installation uses realtime for SIP. The kamailio configuration (attached) was based on the reference at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb but has been heavily modified. Currently asteri

[SR-Users] jsSIP client disconnects after a while when using Kamailio websocket

2014-04-11 Thread Alex Villací­s Lasso
I am having this issue when using kamailio-4.1.2 (after patching the segfault bug). The kamailio is configurated as a frontend for asterisk. When using UDP for SIP, this works correctly. However, I am now setting up websockets (no TLS yet), and I am getting a disconnect after an interval. I am get

Re: [SR-Users] Segfault when trying to send MESSAGE through websocket with jsSIP

2014-04-09 Thread Alex Villací­s Lasso
El 09/04/14 16:17, Daniel-Constantin Mierla escribió: The crash was not related to websocket at all, but to usage of uac_reg_lookup() when you don't set reg_db_url parameter for uac module. So, my question is why are you using uac_reg_lookup()? Because it is useless if you dont set reg_db_url a

Re: [SR-Users] Segfault when trying to send MESSAGE through websocket with jsSIP

2014-04-09 Thread Alex Villací­s Lasso
le is initialized only when reg_db_url is set, otherwise there is no source of user profile to do registration for, and no point to do uac_reg_lookup(). I will add safety checks for this case. Cheers, Daniel On 09/04/14 18:30, Alex Villací­s Lasso wrote: El 09/04/14 03:11, Daniel-Constantin Mierla esc

Re: [SR-Users] Segfault when trying to send MESSAGE through websocket with jsSIP

2014-04-09 Thread Alex Villací­s Lasso
param("uac", "reg_db_url", DBASTURL) # uacreg table is actually a view in DBASTURL #modparam("uac", "reg_contact_addr", "127.0.0.1") Cheers, Daniel On 09/04/14 00:23, Alex Villací­s Lasso wrote: El 04/04/14 16:26, Alex Villací­s Lasso escribió: I am c

Re: [SR-Users] Segfault when trying to send MESSAGE through websocket with jsSIP

2014-04-08 Thread Alex Villací­s Lasso
El 04/04/14 16:26, Alex Villací­s Lasso escribió: I am currently trying to replace a pure-Asterisk implementation of SIP messaging through Websockets, with a Kamailio-4.1.2-based implementation. However, when I try to send a message with jsSIP, Kamailio crashes: Program terminated with signal

[SR-Users] Segfault when trying to send MESSAGE through websocket with jsSIP

2014-04-04 Thread Alex Villací­s Lasso
I am currently trying to replace a pure-Asterisk implementation of SIP messaging through Websockets, with a Kamailio-4.1.2-based implementation. However, when I try to send a message with jsSIP, Kamailio crashes: Program terminated with signal 11, Segmentation fault. #0 0x7f0e5cf31be3 in r

Re: [SR-Users] Memory leak when using db_unixodbc, help tracking it down (try 2)

2014-03-26 Thread Alex Villací­s Lasso
El 24/03/14 09:46, Alex Villací­s Lasso escribió: El 24/03/14 04:17, Daniel-Constantin Mierla escribió: Hello, I found some cases when variables were not freed, but I cannot test as I am not using unixodbc. Can you cherry pick the patch: - http://git.sip-router.org/cgi-bin/gitweb.cgi/sip

Re: [SR-Users] Memory leak when using db_unixodbc, help tracking it down (try 2)

2014-03-24 Thread Alex Villací­s Lasso
El 24/03/14 04:17, Daniel-Constantin Mierla escribió: Hello, I found some cases when variables were not freed, but I cannot test as I am not using unixodbc. Can you cherry pick the patch: - http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=edc78dfb148c22f0d256485193bbdb0185

[SR-Users] Memory leak when using db_unixodbc, help tracking it down (try 2)

2014-03-21 Thread Alex Villací­s Lasso
I have a CentOS 6 installation with the following packages installed from the RPM build service from Kamailio: kamailio-unixodbc-4.1.2-1.1.x86_64 kamailio-4.1.2-1.1.x86_64 kamailio-presence-4.1.2-1.1.x86_64 kamailio-utils-4.1.2-1.1.x86_64 I am also using db_unixodbc for all database accesses (w

[SR-Users] How to encode domain in username for Asterisk REGISTER forwarding (was: How to configure Kamailio + Asterisk (on same server) to route between several disjoint networks?)

2014-03-08 Thread Alex Villací­s Lasso
On the previous issue (disjoint networks), I eventually settled for binding Asterisk on localhost:5080, and using rtpproxy to route media to the networks as required. The only snag is that I had to write a script to rewrite the Kamailio configuration in order to take current IPs into account. No

Re: [SR-Users] How to configure Kamailio + Asterisk (on same server) to route between several disjoint networks?

2014-02-27 Thread Alex Villací­s Lasso
El 26/02/14 11:39, Alex Villací­s Lasso escribió: El 26/02/14 05:25, Klaus Darilion escribió: Puh, too many questions in one email. I am sorry about that. I wanted to provide as much useful information as possible on my email, including what I had observed, and what I have already tried. After

Re: [SR-Users] How to configure Kamailio + Asterisk (on same server) to route between several disjoint networks?

2014-02-26 Thread Alex Villací­s Lasso
) asterisk-only scenario, media is routed between the test networks through asterisk when it does all of the SIP negotiation itself. So, what setup have you choosen? Then we can think about problems. regards Klaus Am 25.02.2014 23:31, schrieb Alex Villací­s Lasso: As part of a project, I h

[SR-Users] How to configure Kamailio + Asterisk (on same server) to route between several disjoint networks?

2014-02-25 Thread Alex Villací­s Lasso
As part of a project, I have installed a CentOS 6 test system (a virtual machine) with Asterisk 11.7.0 and Kamailio 4.1.1 downloaded from http://download.opensuse.org/repositories/home:/kamailio:/telephony/CentOS_CentOS-6/x86_64/. I am trying to setup a combination of Kamailio and Asterisk that wi