sr-users October 2007

sr-users@lists.kamailio.org

130 participants
285 discussions

by Daniel

Hi All, I'd like to extend CPL functionality adding a new switch element. How can manage it ? I think I'd change the "cpl.dtd" file and "cpl-c" module, it's right? Thanks, Daniel -- Daniel Grotti ________________________ e-mail : d.grotti(a)gmail.com

16 years, 8 months

Re: [Serusers] SER + Linksys SPA942 + Call transfer

by roayu

Hi Klaus! The UA does not received the REFER sent by SPA942. What should I need to configure to ensure that the UA receive the REFER from the transfer agent(SPA942)? Thanks. Cheers, Roa Yu -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of kgfleischmann(a)t-online.de Sent: Friday, October 12, 2007 2:22 PM To: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer The NOTIFY is always sent from the UA, which received the REFER. SER only relays it. Did you check this? Cheers Klaus roayu wrote: > Hi there! > > After SPA942(transfer agent) sent REFER to SER, SER does not send a NOTIFY > to SPA942(transfer agent). What's causes this? Can anyone tell me. > > Thanks. > > Cheers, > Roa Yu :) > > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu > Sent: Thursday, October 11, 2007 4:34 PM > To: Atle Samuelsen > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer > > > Hi Atle, > > Let me make my scenario clear on you. > > I have 2 softphone and 1 IP phone(SPA942). I would like to call from > Softphone A to SPA942, then use SPA942 transfer the call to Softphone B. > But, when I do so, the connection still remained on between SPA942 and > Softphone B. > > I managed to get Softphone A onHold and the SPA942 managed to call to > Softphone B. Once I press on the 'xfer' button after calling Softphone B, A > still onHold and no SIP signal to unHold A and to terminate B. > > If I would like to configure the ser.cfg file, which part that I need to > configure? And how do I do that? > > Thanks. > > Cheers, > Roa Yu > > > -----Original Message----- > From: Atle Samuelsen [mailto:clona@cyberhouse.no] > Sent: Thursday, October 11, 2007 3:19 PM > To: roayu > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer > > > Hi Roa,i > * roayu <roayu(a)ctisys.net> [071011 04:01]: >> Hi, there! >> >> I have some questions and help. >> 1) I would like to know can call transfer being done when the >> connection is P2P? When I tried to use relay (mediaproxy) to do the call >> transfer, it's able to transfer the call to the other party whereas when > the >> condition is P2P, it just can't pass the correct signal. > > I'm proberbly only tierd (9:15 am here) but P2P? Can you provide the > signalling and ser.cfg so we can understand what you are trying to > establish? > > I personally think you are trying to do a call transfer from one ua to a > other, where the original call went true mediaproxy, but you do not want > the "new" transferd call to go true it. > >> 2) Can SER support SPA942 ? Or is there some other settings that I need >> to configure on SER ? > > SPA942, SPA962, SPA2102, yea.. all Linksys SPA products work like a > charm with SER (Who has a 942 as his primary phone these days) > > > Best Regards > > ATle >> Thanks. >> >> Cheers, >> Roa Yu >> >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > _______________________________________________ > Serusers mailing list > Serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

16 years, 8 months

[Serusers] Error to start ser

by Angela Gao

Hello, all, I download ser-2.0.0-rc1_src.tar.gz source file and compile it in my own os, and it is: Linux version 2.6.9-34.ELsmp ( bhcompile(a)hs20-bc1-7.build.redhat.com) (gcc version 3.4.5 20051201 (Red Hat 3.4.5-2)). The compile cmd is : make group_include="standard mysql" install I also get ser.buildsystem.latest.tar.gz and create my ser.cfg, no error during the compile. but error is thrown when run ser with cmd: ser -f /home/linus/gao/usr/local/etc/ser.cfg, error will be throued as below: [linus@hp380mf2 sbin]$ ser -f /home/linus/gao/usr/local/etc/ser.cfg ERROR: bad config file (1 errors) ser.cfg is attached in this mail why? Is there anyone can help me fix it? It would be very nice of you, thank you. -- Best regard, Angela

16 years, 8 months

Re: [Serusers] SER + Linksys SPA942 + Call transfer

by roayu

Hi there! After SPA942(transfer agent) sent REFER to SER, SER does not send a NOTIFY to SPA942(transfer agent). What's causes this? Can anyone tell me. Thanks. Cheers, Roa Yu :) -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of roayu Sent: Thursday, October 11, 2007 4:34 PM To: Atle Samuelsen Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer Hi Atle, Let me make my scenario clear on you. I have 2 softphone and 1 IP phone(SPA942). I would like to call from Softphone A to SPA942, then use SPA942 transfer the call to Softphone B. But, when I do so, the connection still remained on between SPA942 and Softphone B. I managed to get Softphone A onHold and the SPA942 managed to call to Softphone B. Once I press on the 'xfer' button after calling Softphone B, A still onHold and no SIP signal to unHold A and to terminate B. If I would like to configure the ser.cfg file, which part that I need to configure? And how do I do that? Thanks. Cheers, Roa Yu -----Original Message----- From: Atle Samuelsen [mailto:clona@cyberhouse.no] Sent: Thursday, October 11, 2007 3:19 PM To: roayu Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER + Linksys SPA942 + Call transfer Hi Roa,i * roayu <roayu(a)ctisys.net> [071011 04:01]: > Hi, there! > > I have some questions and help. > 1) I would like to know can call transfer being done when the > connection is P2P? When I tried to use relay (mediaproxy) to do the call > transfer, it's able to transfer the call to the other party whereas when the > condition is P2P, it just can't pass the correct signal. I'm proberbly only tierd (9:15 am here) but P2P? Can you provide the signalling and ser.cfg so we can understand what you are trying to establish? I personally think you are trying to do a call transfer from one ua to a other, where the original call went true mediaproxy, but you do not want the "new" transferd call to go true it. > > 2) Can SER support SPA942 ? Or is there some other settings that I need > to configure on SER ? SPA942, SPA962, SPA2102, yea.. all Linksys SPA products work like a charm with SER (Who has a 942 as his primary phone these days) Best Regards ATle > > Thanks. > > Cheers, > Roa Yu > > _______________________________________________ > 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

16 years, 8 months

[OpenSER-Users] RE: openxcap Problem

by Schumann Sebastian

Hi Kevin. Unfortunately not. I read the tickets at openxcap.org but that didn't help me yet. I tried plenty of things, downgrade the libxml2 as recommended there, install the required stuff manually but neither worked out. You think the problem comes from xml schema checking from Proxy? I had actually similar problem on other compilation, but in this case I doubt as all xsd files for presence related AUIDs are brought by the openxcap package itself. I would also appreciate if sb. tried compilation and use on a "more current unstable" Debian system or similar (acc. library version differences) and can give additional hints. How did u proceed and so on - there must be some differences to the installation recommendation on openxcap.org. Best regards Sebastian -----Original Message----- From: kinnan2224(a)hotmail.com [mailto:kinnan2224@hotmail.com] Sent: Thu 10/11/2007 16:24 To: Schumann Sebastian Subject: openxcap Problem Hi, Sebastian. I met the same problem as yours. I am using Ubuntu 7.04 at the moment. Have you fixed that problem? I dont if the problem comes from xml schema checking via http with proxy. Can you give me some help? THANK U Regards, Kevin Schumann Sebastian wrote: > > Dear all > > I have a big problem with openxcap. I don't get it work :( As I saw on > their webpage, this mailing list can be posted for this concern as well. > > I am using version 0.9.3 and acc. the installation file on > http://www.openxcap.org all current dependencies from the Debian > unstable distribution. > > When I start /usr/bin/openxcap, the following error occurs in the > syslog: > > Sep 25 17:01:21 openxcap openxcap[11876]: [-] Log opened. > > Sep 25 17:01:21 openxcap openxcap[11876]: [-] Starting Open XCAP 0.9.3 > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] Traceback (most recent > call last): > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/bin/openxcap", line 56, in ? > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.server > import XCAPServer > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/lib/python2.4/site-packages/xcap/server.py", line 21, in ? > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap import > authentication > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/lib/python2.4/site-packages/xcap/authentication.py", line 21, in ? > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] from xcap.appusage > import getApplicationForURI > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 466, > in ? > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] applications = > {'xcap-caps': XCAPCapabilitiesApplication(), > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File > "/usr/lib/python2.4/site-packages/xcap/appusage/__init__.py", line 64, > in __init__ > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] self.xml_schema = > etree.XMLSchema(xml_schema_doc) > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] File "xmlschema.pxi", > line 67, in etree.XMLSchema.__init__ > > Sep 25 17:01:27 openxcap openxcap[11876]: [-] etree.XMLSchemaParseError: > Document is not valid XML Schema > > > > As I wrote, I followed all instructions from the openxcap webpage but > didn't find any hints how to make it work. > > > > Any help on that would be appreciated. > > > > Best regards > > Sebastian > > > -- > Sebastian Schumann > Diploma Student Architecture and Design Department > > Slovak Telekom, a. s. > T-Com, Operations Unit > Production and Service Division > Innovations and Enterprise Solution Subunit > > Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia > Office: Room 449, Jarabinkova 1, 821 09 Bratislava, Slovakia > > +421 2 588 13332 (tel) > +421 910 643010 (mobile) > +49 175 1925928 (mobile) > > sebastian.schumann(a)t-com.sk > http://www.t-com.sk <http://www.t-com.sk/> > > > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users > > Quoted from: http://www.nabble.com/openxcap-Problem-tf4515559.html#a12879532

16 years, 8 months

[OpenSER-Users] Dumb question.

by Alex Balashov

Is there a 'line extend' / 'line wrap' character that can be used inside the OpenSER config, a la Statement("with extensive parameters",\ "some continuation here"); I am sick of having to write obscenely long and hard-to-read lines in my config, which I have to do mainly because I issue SQL queries. -- Alex Balashov Evariste Systems Web : http://www.evaristesys.com/ Tel : +1-678-954-0670 Direct : +1-678-954-0671

16 years, 8 months

[Serusers] ser response port are different with asterisk

by Jun Yin

I noticed a difference between ser and asterisk. When a phone send a register packet with src port 30001, dst port 5060 and "via:x.x.x.x:32001", ser will response a 200 response with "src port 5060, dst port 30001 and "via:x.x.x.x:32001", but asterisk will response a 200 response with "src port 5060, dst port 32001 and "via:x.x.x.x:32001" Don't know why they make this difference? which one more obey the sip standard? thanks. -- Rgds, Hans Yin Web: homeofhans.homeip.net Email: hansyin(a)gmail.com MSN: hansyin(a)hotmail.com Skype: hans_yin_vancouver

16 years, 8 months

[OpenSER-Users] Presence BLF in OpenSer?

by Iñaki Baz Castillo

Hi, many phones as Thomson S2030 or Grandstream 3000 allow presence BLF. Unfortunatelly I think BLF (Busy Lamp Field) just works with Asterisk and it's not a RFC or a draft. Thomson S2030 allow subscribing to a extension state and it generates this SUBSCRIBE: ------------------------------------------------------------------------------- SUBSCRIBE sip:211@domain.org:5061;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.97:5061;branch=z9hG4bK4293581925314869375 From: <sip:210@domain.org:5050>;tag=c0a80101-167c2d To: <sip:211@domain.org:5061> Call-ID: 8b5c839-c0a80101-d-d5(a)192.168.1.97 CSeq: 1 SUBSCRIBE Max-Forwards: 70 Event: dialog Accept: application/dialog-info+xml Expires: 3600 Contact: <sip:210@192.168.1.97:5061;user=phone> User-Agent: THOMSON ST2030 hw5 fw1.52 00-14-7F-00-68-81 Content-Length: 0 ------------------------------------------------------------------------------- Is there any possibility to use this BLF feature with OpenSer in any way? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

16 years, 8 months

[Serusers] Problem with PSTN calls from Asterisk towards SIP clients registered in SER

by Max Weltz

Hello, I am working on a project in which calls are placed from an Asterisk server towards the PSTN or SIP clients and then connected to another phone of the PSTN or another SIP client. Those calls are placed by a user (caller) through an interface. First a call is placed towards the caller from an extension in Asterisk and once the caller picks up the phone, the second call is placed and both are connected. In a first time all SIP clients were registered to Asterisk and there was no SER on the picture. Everything was working fine. But I have decided to use SER as SIP proxy now. So the SIP clients are registered with SER and Asterisk is still the originating point of all calls and provides the connection to the PSTN. Asterisk and SER are running on the same machine (Asterisk has port 5061 and SER port 5060) and Asterisk is registered to SER: ==sip.conf== [general] port=5061 bindport=5061 bindaddr=0.0.0.0 disallow=all ;allow=gsm allow=ulaw allow=alaw ;context=bogon-calls context=from-sip autocreatepeer=yes register=> asterisk:password@192.168.1.24:5060/maxAS [SERADDRESS] type=friend username=asterisk secret=password host=192.168.1.24 fromdomain=192.168.1.24 [maxAS] type=friend secret=password2 qualify=yes nat=no host=dynamic canreinvite=no context=from-sip Most things work fine: * a user can ask Asterisk to call a SIP client and connect it to another SIP client: Asterisk finds its way to SER and both calls are happily connected. * a user can also ask a call to be placed towards a SIP client and then this call is connected to a PSTN call, no problem either. The problem is when the user wants a call towards the PSTN to be connected to a SIP client. Once the call going towards the PSTN has been answered, Asterisk attempts to contact a SIP client which is registered in SER but SER sees the call coming from "sip:(PSTN phone number of the first call)@hostIP". There is no such client in the database and it is therefore not allowed to connect. In the following example Zap/1-1 is the interface towards the PSTN, 0761111111 the phone number dialled, and 6644 an extension registered with SER. Here is the Asterisk output. > Channel Zap/1-1 was answered. -- Executing Answer("Zap/1-1", "") in new stack -- Executing Goto("Zap/1-1", "from-sip|6644|1") in new stack -- Goto (from-sip,6644,1) -- Executing Answer("Zap/1-1", "") in new stack -- Executing Dial("Zap/1-1", "SIP/6644@SERADDRESS||r") in new stack -- Called 6644@SERADDRESS Oct 10 11:41:16 WARNING[4264]: chan_sip.c:7972 handle_response: Forbidden - wrong password on authentication for INVITE to '"0761111111" <sip:0761111111@192.168.1.24>;tag=as537d1ed7' -- SIP/SERADDRESS-50e7 is circuit-busy Here is the portion of the Asterisk Dialplan that is relevant and the ser.cfg (which comes from http://siprouter.onsip.org/doc/ gettingstarted/): ==extensions.conf== [context1] exten => 666,1,Answer() exten => 666,2,Goto(from-sip,6644,1) [from-sip] exten => _.,1,Answer exten => _.,2,Dial(SIP/${EXTEN}@SERADDRESS,,r) exten => _.,3,Hangup ==ser.cfg== debug=3 fork=no log_stderror=yes listen=192.168.1.24 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:password@192.168.1.26/ser" loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" loadmodule "/usr/lib/ser/modules/uri_db.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql:// ser:password@192.168.1.26/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("192.168.1.24","subscriber")) { www_challenge("192.168.1.24","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("192.168.1.24","subscriber")) { proxy_challenge("192.168.1.24","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } I guess (but I am new to SER) that the idea is to make the call be placed as the SIP user "asterisk" and not as the SIP user "0761111111" but I don't know where to do that (inside asterisk before the SIP call is placed? inside ser.cfg?) or maybe just to skip the authentication for SIP URIs that look like a phone number. I also read in http://siprouter.onsip.org/doc/gettingstarted/ch09.html how to relay calls placed towards the PSTN from SER towards a gateway (Asterisk here) but I think it wouldn't do any good to send back the call to Asterisk. Asterisk would just end up placing another call towards the PSTN and that's not what I want. Any suggestion will be greatly appreciated, Thanks, //Max

16 years, 8 months

Re: [OpenSER-Users] Set up two SIP to PSTN calls and then connect them

by Andreas Granig

> To my knowledge, it’s not possible to use OpenSER alone to do this. > Options include WeSIP (beta software, restrictive licence) or Asterisk > (poor implementation of “peer” identification limits usability). Could > anyone advise what other alternatives there are? Is it possible to just > use OpenSER? What about SEMS? In the source tree, there is "examples/ctd.sh", which relies on the help of the involved UACs (using REFER). For just INVITE, SEMS is a good choice, and we plan to submit our click2dial plugin into the trunk within the next days. Andreas

16 years, 8 months

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sr-users October 2007