October 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

After several hours FS registers ITSP every minute, requires FS restart
by Mario 05 Oct '10

05 Oct '10

I have 2 accounts with an ITSP that with the SIP profile below. After Freeswtich has been up for several hours, especially during my overnight testing, the accounts start to register every minute. This would not be really bad except that incoming calls don't work and I have to restart FS. I have the sofia trace below that also includes 407 Proxy Authentication Required messages but that does not seem to affect anything, the account works. Using the git version from this past Saturday: FreeSWITCH Version 1.0.head (git-f288e3c 2010-10-01 23-35-51 -0400). Thanks, Mario <include> <gateway name="uuid1"> <param name="proxy" value="sip.itsp.com"/> <param name="username" value="uuid1"/> <param name="from-user" value="uuid1"/> <param name="password" value="password"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> </gateway> </include> 2010-10-04 16:36:09.758277 [NOTICE] sofia_reg.c:342 Registering uuid1 send 865 bytes to udp/[216.143.130.36]:5060 at 23:36:09.759509: ------------------------------------------------------------------------ REGISTER sip:sip.itsp.com SIP/2.0 Via: SIP/2.0/UDP 106.88.82.100:5080;rport;branch=z9hG4bKamKeDBQrZ9j3m Max-Forwards: 70 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp> Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768939 REGISTER Contact: <sip:gw+uuid1@106.88.82.100:5080;transport=udp;gw=uuid1> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f288e3c 2010-10-01 23-35-51 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Proxy-Authorization: Digest username="uuid1", realm="rnktel.com", nonce="4caa64a90001484bedc7fa432b7fba4e53832e2474cb5555", algorithm=MD5, uri="sip:sip.itsp.com", response="7290bc4ec3cd32798c74d9ebd62c247e" Content-Length: 0 ------------------------------------------------------------------------ recv 567 bytes from udp/[216.143.130.36]:5060 at 23:36:09.866684: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 106.88.82.100:5080;rport=5080;branch=z9hG4bKamKeDBQrZ9j3m;received=209.143.244.127 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp>;tag=9214955d68c027e4c99a14a397962871.075c Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768939 REGISTER Proxy-Authenticate: Digest realm="rnktel.com", nonce="4caa64e500015288f99aa6d49e6686bbe3e8d9d066275ce8", stale=true Server: OpenSIPS (1.6.3-notls (i386/freebsd)) Content-Length: 0 ------------------------------------------------------------------------ send 865 bytes to udp/[216.143.130.36]:5060 at 23:36:09.867061: ------------------------------------------------------------------------ REGISTER sip:sip.itsp.com SIP/2.0 Via: SIP/2.0/UDP 106.88.82.100:5080;rport;branch=z9hG4bKBXc7e67Uvj9Ng Max-Forwards: 70 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp> Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768940 REGISTER Contact: <sip:gw+uuid1@106.88.82.100:5080;transport=udp;gw=uuid1> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f288e3c 2010-10-01 23-35-51 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Proxy-Authorization: Digest username="uuid1", realm="rnktel.com", nonce="4caa64e500015288f99aa6d49e6686bbe3e8d9d066275ce8", algorithm=MD5, uri="sip:sip.itsp.com", response="ea063d8a376643e491e1c41ce09dcf6e" Content-Length: 0 ------------------------------------------------------------------------ recv 549 bytes from udp/[216.143.130.36]:5060 at 23:36:09.975863: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 106.88.82.100:5080;rport=5080;branch=z9hG4bKBXc7e67Uvj9Ng;received=209.143.244.127 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp>;tag=9214955d68c027e4c99a14a397962871.aa06 Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768940 REGISTER Contact: <sip:gw+uuid1@209.143.244.127:5080;transport=udp;gw=uuid1>;expires=60, <sip:uuid1@209.143.244.127:5061>;expires=279 Server: OpenSIPS (1.6.3-notls (i386/freebsd)) Content-Length: 0 ------------------------------------------------------------------------ 2010-10-04 16:37:09.575144 [NOTICE] sofia_reg.c:342 Registering uuid1 send 865 bytes to udp/[216.143.130.36]:5060 at 23:37:09.623804: ------------------------------------------------------------------------ REGISTER sip:sip.itsp.com SIP/2.0 Via: SIP/2.0/UDP 106.88.82.100:5080;rport;branch=z9hG4bKc65Zg1rZSUZ8B Max-Forwards: 70 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp> Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768941 REGISTER Contact: <sip:gw+uuid1@106.88.82.100:5080;transport=udp;gw=uuid1> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f288e3c 2010-10-01 23-35-51 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Proxy-Authorization: Digest username="uuid1", realm="rnktel.com", nonce="4caa64e500015288f99aa6d49e6686bbe3e8d9d066275ce8", algorithm=MD5, uri="sip:sip.itsp.com", response="ea063d8a376643e491e1c41ce09dcf6e" Content-Length: 0 ------------------------------------------------------------------------ recv 567 bytes from udp/[216.143.130.36]:5060 at 23:37:09.723821: ------------------------------------------------------------------------ SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 106.88.82.100:5080;rport=5080;branch=z9hG4bKc65Zg1rZSUZ8B;received=209.143.244.127 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp>;tag=9214955d68c027e4c99a14a397962871.9bd6 Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768941 REGISTER Proxy-Authenticate: Digest realm="rnktel.com", nonce="4caa652100015c59a0f47e6209ee434ff2e43a548d62af15", stale=true Server: OpenSIPS (1.6.3-notls (i386/freebsd)) Content-Length: 0 ------------------------------------------------------------------------ send 865 bytes to udp/[216.143.130.36]:5060 at 23:37:09.724214: ------------------------------------------------------------------------ REGISTER sip:sip.itsp.com SIP/2.0 Via: SIP/2.0/UDP 106.88.82.100:5080;rport;branch=z9hG4bKDFZrjv92p4NUQ Max-Forwards: 70 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp> Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768942 REGISTER Contact: <sip:gw+uuid1@106.88.82.100:5080;transport=udp;gw=uuid1> Expires: 60 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f288e3c 2010-10-01 23-35-51 -0400 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Proxy-Authorization: Digest username="uuid1", realm="rnktel.com", nonce="4caa652100015c59a0f47e6209ee434ff2e43a548d62af15", algorithm=MD5, uri="sip:sip.itsp.com", response="48be3b8135d84350c145baf028460cfb" Content-Length: 0 ------------------------------------------------------------------------ recv 549 bytes from udp/[216.143.130.36]:5060 at 23:37:09.824242: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 106.88.82.100:5080;rport=5080;branch=z9hG4bKDFZrjv92p4NUQ;received=209.143.244.127 From: <sip:uuid1@sip.itsp.com;transport=udp>;tag=vSDFr2v77m0yc To: <sip:uuid1@sip.itsp.com;transport=udp>;tag=9214955d68c027e4c99a14a397962871.c855 Call-ID: b1ecbbc2-aea4-4f58-98d0-1dad278e24c7 CSeq: 2768942 REGISTER Contact: <sip:gw+uuid1@209.143.244.127:5080;transport=udp;gw=uuid1>;expires=60, <sip:uuid1@209.143.244.127:5061>;expires=219 Server: OpenSIPS (1.6.3-notls (i386/freebsd)) Content-Length: 0

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Substitution of mod_g729 by mod_com_g729 causes core dump
by Jan Riedinger 04 Oct '10

04 Oct '10

I just tried to subsitute mod_g729 by mod_com_g729 by using an appropriate moduls.conf.xml. After a restart of Freeswitch, it crahses, when the first inbound call gets processed: freeswitch(a)sns-fs1.sns.eu> 2010-10-04 12:32:27.064152 [NOTICE] switch_channel.c:779 New Channel sofia/external/xyz(a)a.b.c.d [8e94789e-5a8d-4088-8ad1-cd5fb2be0c4f] freeswitch: src/switch_core_memory.c:448: switch_core_perform_alloc: Assertion `pool != ((void *)0)' failed. Aborted (core dumped) There seems to be no error, when mod_com_g729 gets loaded: 2010-10-04 12:32:21.856829 [INFO] mod_com_g729.c:243 Permitted G.729AB channels: 20 2010-10-04 12:32:21.856920 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 10ms 8000bps 2010-10-04 12:32:21.856925 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 20ms 8000bps 2010-10-04 12:32:21.856928 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 30ms 8000bps 2010-10-04 12:32:21.856933 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 40ms 8000bps 2010-10-04 12:32:21.856937 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 50ms 8000bps 2010-10-04 12:32:21.856943 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 60ms 8000bps 2010-10-04 12:32:21.856947 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 70ms 8000bps 2010-10-04 12:32:21.856974 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 80ms 8000bps 2010-10-04 12:32:21.856978 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 90ms 8000bps 2010-10-04 12:32:21.856983 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 100ms 8000bps 2010-10-04 12:32:21.856988 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 110ms 8000bps 2010-10-04 12:32:21.856993 [NOTICE] switch_loadable_module.c:185 Adding Codec G729 18 G.729 8000hz 120ms 8000bps 2010-10-04 12:32:21.857144 [NOTICE] switch_loadable_module.c:274 Adding API Function 'g729_count' 2010-10-04 12:32:21.857175 [NOTICE] switch_loadable_module.c:274 Adding API Function 'g729_used' 2010-10-04 12:32:21.857199 [NOTICE] switch_loadable_module.c:274 Adding API Function 'g729_available' 2010-10-04 12:32:21.857229 [NOTICE] switch_loadable_module.c:274 Adding API Function 'g729_info' I'm using FreeSWITCH Version 1.0.head (git-758f383 2010-09-30 13-30-36 -0700) If I use mod_g729 instead of mod_com_g729 Freeswitch is running stable. Does anybody have any ideas how to debug this issue? Thank you in advance Jan -- Jan Riedinger Phone : +49-30-39 73 19 66 Dipl.-Inf. | Managing Director Fax : +49-30-39 73 19 64 E-Mail: riedinger(a)sns.eu SNS Consult GmbH ICQ : 163-237-041 Südwestkorso 49a MSN : jan(a)sns-consult.de 14197 Berlin GERMANY Skype : Jan Riedinger AG Charlottenburg - HRB 71973

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voicemail.tpl vs notify-voicemail.tpl
by Mario 04 Oct '10

04 Oct '10

I now have email working but could not find info about the differences between voicemail.tpl and notify-voicemail.tpl. They look the same and I wonder when each are used. I see they are referenced in voicemail.xml. Thanks. Mario

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FreeSWITCH HA
by Kane, Michael (mkane02) 04 Oct '10

04 Oct '10

Will FreeSWITCH HA work if bypass_media = true? Thanks Mike

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Codec question: about the G729 codec
by joy this 04 Oct '10

04 Oct '10

Dear all: I have some questions about the G729 codec. I will be appreciated if someone could help me. 1. If I want to modify the G729 codec, I must compile the module and substitute for the original one. Is it correct? 2. It seems that there is an authentication server for G729. Should I need authentication when I use the modified G729 codec? 3. If I want to add my codec which is not in core, how to do it? Thanks for helping.

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sip to spa3102 fallback
by Eric Masson 04 Oct '10

04 Oct '10

Hello, I have two sip providers configuration files in $(fs_conf}/conf/diaplan/default : - 00_keyyo.net.xml <include> <extension name="keyyo" continue="false"> <condition field="destination_number" expression = "^(.*)$"> <action application="set" data="effective_caller_id_name=**MyID**"/> <action application="set" data="effective_caller_id_number=**Myphonenumber"/> <action application="bridge" data="sofia/gateway/keyyo.net/$1"/> </condition> </extension> </include> - 00_spa3102.xml <include> <extension name="To PSTN"> <condition field="destination_number" expression="^(.*)$"> <action application="bridge" data="sofia/internal/${destination_number}@spafenmsf:5061" /> </condition> </extension> </include> I'd like to implement the following scenario for outbound calls : - dial using "keyyo" if available - fallback to "To PSTN" if keyyo isn't available (not registered, already used, for example) I've googled for freeswitch failover setups but didn't find anything that could have enlightened me (I'm kind of new in fs). Where could I find docs to achieve the exposed goal please ? Kind Regards Eric Masson

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Accessing wanpipe from openVZ container
by Russell Mosemann 04 Oct '10

04 Oct '10

We are using openVZ with FreeSWITCH running in a container. In another situation, I successfully installed DAHDI in the host node, exported the devices to the container and installed libpri in the container. FS could access the card without any problems. We are moving to a Sangoma A101D and an A200. I have installed the wanpipe drivers in the host node (make freetdm and make install), exported the devices to the container and installed the signaling in the container (make install_pri). However, that is not quite enough for the container, because mod_freetdm looks for libsangoma, which is in the host node. wancfg_fs does too much, because it attempts to configure wanrouter and FS at the same time. I've looked through the FS wiki and the FS section of the Sangoma wiki, but all of the directions are for a combined installation. Is there a set of steps to install the wanpipe drivers separately from the signaling and do the individual configuration for each? Thanks. Russell

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openzap error "RECOVERY_ON_TIMER_EXPIRE"
by ovvenkat 04 Oct '10

04 Oct '10

Hi to all, I am using Sangoma A101 card and openzap to make outbound call. When I am trying to originate call, getting blow error 2010-10-04 17:34:16.606653 [ERR] switch_ivr_originate.c:2648 Cannot create outgoing channel of type [openzap] cause: [RECOVERY_ON_TIMER_EXPIRE] 2010-10-04 17:34:16.606653 [DEBUG] switch_ivr_originate.c:3456 Originate Resulted in Error Cause: 102 [RECOVERY_ON_TIMER_EXPIRE] Any one come across this problem, what is this mean and how to rectify the same ? Thanks in advance. Regards, Venkat. -- If you have come to help me, you are wasting your time. If you have come to because your liberation is bound up in mine, we can work together. Regards Venkatesan OV.

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Re: [Freeswitch-users] Setting a domain name in the Contact header of an INVITE
by Neill Wilkinson 04 Oct '10

04 Oct '10

Gavin, this kinda thing works for me: <action application="export" data="sip_contact_user=${sip_from_user}" /> Contact user seems to overwrite the whole field, so something like: <action application="export" data="sip_contact_user=gw+test@test.domain.co.uk:5060;transport=udp;gw=text" /> Neill.....;o) Aeonvista Ltd <http://aeonvista.blogspot.com> *Opening Up New Ideas* ---------- Forwarded message ---------- > From: Gavin Henry <gavin.henry(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 4 Oct 2010 13:52:08 +0100 > Subject: [Freeswitch-users] Setting a domain name in the Contact header of > an INVITE > Hi, > > Is this possible? > > Contact: <sip:gw+test@test.domain.co.uk:5060;transport=udp;gw=text> > > instead of: > > Contact: <sip:gw+test@FS_PUBLIC_IP:5060;transport=udp;gw=test> > > > > I've tried exporting sip_contact_host and domain_name etc. > > sip_contact_params only adds string literals to this header. > > Cheers. > > -- > http://www.suretecsystems.com/services/openldap/ > http://www.surevoip.co.uk > > > >

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Some weird problems with Polycom phones.
by Yehavi Bourvine 04 Oct '10

04 Oct '10

Hello, I have a few oddities with Polycom phones which I just have documented at the Wiki. I append them here also for those who might tackle them. The following issues have opened tickets with Polycom's support: - *Dial tone on a shared line*: Suppose you have an SLA extension defined on more than one phone. You answer on one phone, put it on hold, go to the other phone and press the blinking line in order to have the call there. If you press this line and within 1 second pull the handset - you hear a dial tone at the background of the session. If you wait more than 1 second or first pull the handset (and then press the line button) - all is ok. - *Dial using the second line*: If you have a phone with more than one extension defined on it, it will will use the first line for outgoing calls by default (unless of course you select a specific line). However, there is one special case: you dial on-hook and then press the speaker button: - The call is initiated on the first line. - Immediately after getting "Trying" the phone sends CANCEL. - The phone originates the call successfully over the second line. Regards, __Yehavi:

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