Duplicate gateways - how to use multiple user IDs?
In this case you have to name each gateway unique names and use the proxy field to specify the hostname to the proxy. /b On Sep 20, 2010, at 10:45 AM, Mario wrote:
I have more than 1 account at an ITSP, I put both on sip_profiles/external. Both work but only one at a time. At startup I get ignoring duplicate gateway. I looked at the new book (very good) and examples, wiki, etc. but could not find a way to do this. This is a major problem. Anyone know how to do this? Thanks.
-- Mario _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Thanks very much that looks to be working! I am new at FreeSwitch and have read the book (very good), wiki, etc. I am working through my issues and this first post is very encouraging! Thanks to others that responded! Mario On 09/20/2010 09:04 AM, Brian West wrote:
In this case you have to name each gateway unique names and use the proxy field to specify the hostname to the proxy.
/b
On Sep 20, 2010, at 10:45 AM, Mario wrote:
I have more than 1 account at an ITSP, I put both on sip_profiles/external. Both work but only one at a time. At startup I get ignoring duplicate gateway. I looked at the new book (very good) and examples, wiki, etc. but could not find a way to do this. This is a major problem. Anyone know how to do this? Thanks.
-- Mario _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- *Mario*
Although I have the 2 accounts now registering with the same ITSP, I can't get a dial out due to invalid gateway message even though I changed the dialplan and tried many combinations. Now that I have the 2 accounts registered using proxy, how can I get the outbound diaplan to direct to the correct gateway? I thought it was as simple as changing the name to match then gateway name. Thanks! 2010-09-20 10:38:07.974744 [ERR] mod_sofia.c:3690 Invalid Gateway 2010-09-20 10:38:07.974744 [NOTICE] mod_sofia.c:4012 Close Channel N/A [CS_NEW] 2010-09-20 10:38:07.974744 [ERR] switch_ivr_originate.c:2632 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2010-09-20 10:38:07.974744 [INFO] mod_dptools.c:2393 Originate Failed. Cause: INVALID_NUMBER_FORMAT 2010-09-20 10:38:07.974744 [NOTICE] mod_dptools.c:2456 Hangup sofia/internal/200@10.246.1.37 [CS_EXECUTE] [INVALID_NUMBER_FORMAT] SIPS: <gateway name="a1.sip.itsp.com"> <!-- <param name="proxy" value="sip.itsp.com"/> --> <param name="username" value="userid1"/> <param name="from-user" value="userid1"/> <param name="password" value="password1"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> </gateway> <gateway name="a2.sip.itsp.com"> <!-- <param name="proxy" value="sip.itsp.com"/> --> <param name="username" value="userid2"/> <param name="from-user" value="userid2"/> <param name="password" value="password"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> </gateway> DIALPLANS: <extension name="local.a1.sip.voipyourlife.com"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^8(\d{7})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/a1.sip.itsp.com/1${default_areacode}$1@a1.sip.itsp.com"/> </condition> </extension> <extension name="local.a2.sip.voipyourlife.com"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^8(\d{7})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/a2.sip.itsp.com/1${default_areacode}$2@a1.sip.itsp.com"/> </condition> </extension> On 09/20/10 09:34, Mario wrote:
Thanks very much that looks to be working! I am new at FreeSwitch and have read the book (very good), wiki, etc. I am working through my issues and this first post is very encouraging! Thanks to others that responded! Mario
On 09/20/2010 09:04 AM, Brian West wrote:
In this case you have to name each gateway unique names and use the proxy field to specify the hostname to the proxy.
/b
On Sep 20, 2010, at 10:45 AM, Mario wrote:
I have more than 1 account at an ITSP, I put both on sip_profiles/external. Both work but only one at a time. At startup I get ignoring duplicate gateway. I looked at the new book (very good) and examples, wiki, etc. but could not find a way to do this. This is a major problem. Anyone know how to do this? Thanks.
-- Mario _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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Chances are your provider needs a plus sign or some other format enable the sip trace by doing "sofia profile external siptrace on" and place a call. Lets see what your sending. /b On Sep 20, 2010, at 12:48 PM, Mario wrote:
2010-09-20 10:38:07.974744 [ERR] switch_ivr_originate.c:2632 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
I used the trace but it never got far enough to dial. I figured out the problem. The dial plan bridge application line must use the real itsp name at the end, not the proxy name. It looks ok now. Thanks. Now, om to why I can't connect dial out to itsp2... This line: <action application="bridge" data="sofia/gateway/a1.sipt.itsp.com/1${default_areacode}$1@a1.sip.itsp.com"/> Should be: <action application="bridge" data="sofia/gateway/a1.sipt.itsp.com/1${default_areacode}$1@sip.itsp.com"/> On 09/20/10 11:01, Brian West wrote:
Chances are your provider needs a plus sign or some other format enable the sip trace by doing "sofia profile external siptrace on" and place a call. Lets see what your sending.
/b
On Sep 20, 2010, at 12:48 PM, Mario wrote:
2010-09-20 10:38:07.974744 [ERR] switch_ivr_originate.c:2632 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT]
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When I dial out to voipyourlife I get a fast busy signal no matter what number I dial. The messages and trace don't help me. Web searches on freeswitch and viopyourlife/voipyourphone yielded no results. Thanks very much! sofia profile external siptrace on 2010-09-20 11:47:21.170623 [NOTICE] switch_channel.c:779 New Channel sofia/internal/200@10.246.1.37 [2d5efef1-c8ee-4818-8563-937f62d04543] 2010-09-20 11:47:21.318116 [INFO] mod_dialplan_xml.c:331 Processing 200 <200>->814181234567 in context default 2010-09-20 11:47:21.328251 [NOTICE] switch_channel.c:779 New Channel sofia/external/14181234567@sip.voipyourlife.com [7e8f01ae-112b-495a-b091-870f5304f51e] 2010-09-20 11:47:24.100398 [NOTICE] sofia.c:5051 Hangup sofia/external/14181234567@sip.voipyourlife.com [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2010-09-20 11:47:24.100398 [INFO] mod_dptools.c:2393 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2010-09-20 11:47:24.100398 [NOTICE] mod_dptools.c:2456 Hangup sofia/internal/200@10.246.1.37 [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2010-09-20 11:47:24.100398 [NOTICE] switch_core_session.c:1228 Session 4 (sofia/external/14181234567@sip.voipyourlife.com) Ended 2010-09-20 11:47:24.100398 [NOTICE] switch_core_session.c:1230 Close Channel sofia/external/14181234567@sip.voipyourlife.com [CS_DESTROY] 2010-09-20 11:47:24.212365 [NOTICE] switch_core_session.c:1228 Session 3 (sofia/internal/200@10.246.1.37) Ended 2010-09-20 11:47:24.212365 [NOTICE] switch_core_session.c:1230 Close Channel sofia/internal/200@10.246.1.37 [CS_DESTROY] recv 650 bytes from udp/[216.143.130.36]:5060 at 18:48:24.027473: ------------------------------------------------------------------------ NOTIFY sip:gw+a1@99.39.148.147:5080;transport=udp;gw=a1 SIP/2.0 Max-Forwards: 70 Record-Route: <sip:216.143.130.36;lr=on> Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bK444b.67fa5084.0 Via: SIP/2.0/UDP 216.143.130.47:60853;received=216.143.130.47;branch=z9hG4bK.20dfac02;rport=60853;alias From: "Unknown" <sip:Unknown@216.143.130.36> To: <sip:a1@216.143.130.36> Contact: <sip:Unknown@216.143.130.36> Call-ID: 718111ed3c1ed2fb3Rc2Qb6Qr4Nv8Ve8@216.143.130.47 CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: message-summary Content-Type: application/simple-message-summary Content-Length: 36 Messages-Waiting: no Voicemail: 0/0 ------------------------------------------------------------------------ send 738 bytes to udp/[216.143.130.36]:5060 at 18:48:24.028966: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 216.143.130.36;branch=z9hG4bK444b.67fa5084.0 Via: SIP/2.0/UDP 216.143.130.47:60853;received=216.143.130.47;branch=z9hG4bK.20dfac02;rport=60853;alias Record-Route: <sip:216.143.130.36;lr=on> From: "Unknown" <sip:Unknown@216.143.130.36> To: <sip:a1@216.143.130.36>;tag=vrU23ac8SFaFK Call-ID: 718111ed3c1ed2fb3Rc2Qb6Qr4Nv8Ve8@216.143.130.47 CSeq: 102 NOTIFY Contact: <sip:gw+a1@99.39.148.147:5080;transport=udp;gw=a1> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-19dc1ab 2010-09-16 20-08-30 +0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Length: 0 ------------------------------------------------------------------------
You should perhaps use different names. David Ponzone Direction Technique email: david.ponzone@ipeva.fr tel: 01 74 03 18 97 gsm: 06 66 98 76 34 Service Client IPeva tel: 0811 46 26 26 www.ipeva.fr - www.ipeva-studio.com Ce message et toutes les pièces jointes sont confidentiels et établis à l'intention exclusive de ses destinataires. Toute utilisation ou diffusion non autorisée est interdite. Tout message électronique est susceptible d'altération. IPeva décline toute responsabilité au titre de ce message s'il a été altéré, déformé ou falsifié. Si vous n'êtes pas destinataire de ce message, merci de le détruire immédiatement et d'avertir l'expéditeur. Le 20/09/2010 à 17:45, Mario a écrit :
I have more than 1 account at an ITSP, I put both on sip_profiles/ external. Both work but only one at a time. At startup I get ignoring duplicate gateway. I looked at the new book (very good) and examples, wiki, etc. but could not find a way to do this. This is a major problem. Anyone know how to do this? Thanks.
-- Mario _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Have you tried making sure your gateway tag is different for the other account? <gateway name="SOMETHING_ELSE"> Anthony C On Mon, 2010-09-20 at 08:45 -0700, Mario wrote:
I have more than 1 account at an ITSP, I put both on sip_profiles/external. Both work but only one at a time. At startup I get ignoring duplicate gateway. I looked at the new book (very good) and examples, wiki, etc. but could not find a way to do this. This is a major problem. Anyone know how to do this? Thanks.
-- Mario
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
participants (4)
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Anthony Cosgrove
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Brian West
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David Ponzone
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Mario