- FreeSWITCH-users - FreeSWITCH Mailing Lists

in bypass_media mode, 2833 payload type modified by freeswitch!
by qingquan luo 13 Jun '11

13 Jun '11

Hi All, I use freeswitch to bridge one incoming call to other target phone number. I use bypass_media mode. So the rtp is not go through the freeswitch, By I notice that When LegB reply 183 or 200 SDP message. Freeswitch change it 2833 telephone-event payload type. and forwarding the message to caller. This make the caller use wrong dtmf payload type to send dtmf What wrong with that? How can fix it? Any information or help is welcome. Thanks Best Regards Qingquan

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Sangoma Card D100
by Gustavo Espeche 13 Jun '11

13 Jun '11

Hi Some one can do this card work, we was buy it 2 week ago and nor Sangoma support can't it work, the card don't do transcoding, if someone can do it work please let us know. -- Gustavo Espeche EasyIpCall S.R.L. www.easyipcall.com Bv Mitre 517 24° E Cordoba - Argentina Te: +54 - 351 - 4280633

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SIP Option Method
by Andy Ayers 13 Jun '11

13 Jun '11

Hi folks, I've been asked by my telecoms provider if my switch supports the SIP OPTION METHOD. I'm running freeswitch 1.0.6 on a debian linux server. Can you confirm if this is supported or not? Many thanks Andy

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session:read() not timeouting
by David Ponzone 12 Jun '11

12 Jun '11

All, When using session:read() in a LUA script called from dialplan, the interdigit timeout works fine on my test box (with GIT from end of May, and a 1000hz timer). On my prod box, the timeout does not trigger the end of the read app. The terminating character is the only way to end the read() normally. The prod box is using a GIT from December 2010, and a bad timer (250Hz). Should I suppose the issue comes from the timer or from the old version on the prod box (I am upgrading ASAP) ? Thanks David Ponzone Direction Technique email: david.ponzone(a)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.

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g729 licenses are not valid anymore after upgrade
by David Ponzone 12 Jun '11

12 Jun '11

I just upgraded from a 6 months GIT version to current, and the latest mod_com_g729 does not seem to accept my G729 licenses anymore. The error message is: Unrecognized resource G729A/0 If I revert back to the previous version (with mod_com_g729 158), it works fine. Is there something that was changed in the licenses ? Thanks David Ponzone Direction Technique email: david.ponzone(a)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.

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Sofia invite issues
by Kurtis Heimerl 12 Jun '11

12 Jun '11

Hello Freeswitch-users! I'm currently writing a SIP middlebox that intercepts sip messages and changes their username. The reasons for this are complicated and somewhat outside of the scope of this discussion. However, I've come upon a very strange issue: when making a phone-to-phone call across freeswitch; it is connecting the wrong user. I'm fairly well convinced this is a bug, but I thought I'd send the issue here and see if it's anything I'm obviously doing wrong. Basically, FS sees a SIP message from a registered UA (call it A) inviting another registered user to a call (B). This is acked correctly (a TRYING message). FS then responds by inviting B into a call with some OTHER user C. C is not mentioned at any point in the initial SIP messages (as verified by wireshark). Even more strangely, if I remove user C from FS (by removing their config file) FS responds to the invite by inviting B into a call with itself. In each case, the appropriate from header in the invite should be the original caller A. It's worth noting that I am not monkeying with the RTP packets at all, but my understanding is that SIP signalling shouldn't be affected by that. My general guess is that I'm messing up the naming somehow, and FS is running an algorithm to guess at who the call originator is. However, the naming must be roughly correct; Asterisk is able to handle this call just fine. I've included a sip trace of the second situation (A calls B, FS invites B to a call with itself) For the record A:1300 B:1301 C:IMSI641104878332498 REGISTER MESSAGE (Which works fine) REGISTER sip:192.168.1.144 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347 Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347 From: 1300 <sip:1300@192.168.1.144>;tag=ftoui To: 1300 <sip:1300@192.168.1.144> Call-ID: 1032827938(a)192.168.1.144 CSeq: 91 REGISTER Contact: <sip:1300@192.168.1.144:5063>;expires=7200 Max-Forwards: 70 Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347 Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK13347 From: 1300 <sip:1300@192.168.1.144>;tag=ftoui To: 1300 <sip:1300@192.168.1.144>;tag=jU64NXypQc57F Call-ID: 1032827938(a)192.168.1.144 CSeq: 91 REGISTER Contact: <sip:1300@192.168.1.144:5063>;expires=7200 Date: Wed, 08 Jun 2011 21:02:04 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 SIP TRACE (A -> B, B -> B response) INITIAL INVITE INVITE sip:1301@192.168.1.144 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974 From: 1300 <sip:1300@192.168.1.144>;tag=bgdqx To: <sip:1301@192.168.1.144> Call-ID: 1817795092(a)192.168.1.144 CSeq: 485 INVITE Contact: <sip:1300@192.168.1.144:5063>;expires=3600 Content-Type: application/sdp Max-Forwards: 70 Content-Length: 143 INVITE ACK SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.144:5063;branch=z9hG4bK77974 From: 1300 <sip:1300@192.168.1.144>;tag=bgdqx To: <sip:1301@192.168.1.144> Call-ID: 1817795092(a)192.168.1.144 CSeq: 485 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700 Content-Length: 0 FS INVITE (Note the from address being wrong) INVITE sip:1301@192.168.1.144:5063 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.144;rport;branch=z9hG4bKeeQvcyZ70SDUg Max-Forwards: 69 From: "1301" <sip:1301@192.168.1.144>;tag=rHyS0Z3B61arN To: <sip:1301@192.168.1.144:5063> Call-ID: 84df47f9-0cb7-122f-13b5-5cff350d9de5 CSeq: 13447852 INVITE Contact: <sip:mod_sofia@192.168.1.144:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-71d64e6 2011-06-07 22-30-16 -0700 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 315 X-FS-Support: update_display Remote-Party-ID: "1301" <sip:1301@192.168.1.144>;party=calling;screen=yes;privacy=off USER CONFIGURATIONS 1300.xml <include> <user id="1300" cidr="192.168.1.0/24,127.0.0.1/32"> <params> <param name="accept-blind-reg" value="true"/> <param name="accept-blind-auth" value="true"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1300"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="1300"/> <variable name="effective_caller_id_number" value="1300"/> <variable name="outbound_caller_id_name" value="1300"/> <variable name="outbound_caller_id_number" value="1300"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> 1301.xml <include> <user id="1301" cidr="192.168.1.0/24,127.0.0.1/32"> <params> <param name="accept-blind-reg" value="true"/> <param name="accept-blind-auth" value="true"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1301"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="1301"/> <variable name="effective_caller_id_number" value="1301"/> <variable name="outbound_caller_id_name" value="1301"/> <variable name="outbound_caller_id_number" value="1301"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> IMSI641104878332498.xml <include> <user id="IMSI641104878332498" cidr="192.168.1.0/24,127.0.0.1/32" number-alias="1302"> <params> <param name="accept-blind-reg" value="true"/> <param name="accept-blind-auth" value="true"/> <param name="vm-password" value="1000"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1302"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="IMSI641104878332498"/> <variable name="effective_caller_id_number" value="1302"/> <variable name="outbound_caller_id_name" value="IMSI641104878332498"/> <variable name="outbound_caller_id_number" value="1302"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> DIALPLAN <extension name="Local Numbers"> <condition field="destination_number" expression="^130[0-9]$">  <action application="bridge" data="user/${destination_number}"/> </condition> </extension>

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Codec/CDR Error
by Gustavo Espeche 12 Jun '11

12 Jun '11

Hi we found an error in freeswitch 2011-06-12 10:45:35.662591 [ERR] sofia.c:5323 Codec Error! v=0 o=Argentina-2 1307890456 1307890458 IN IP4 200.49.30.68 s=sip call c=IN IP4 200.49.30.69 t=0 0 m=audio 42410 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 -------------------------------------------- 2011-06-12 10:48:28.322557 [ERR] sofia.c:5323 Codec Error! v=0 o=Sonus_UAC 19969 23951 IN IP4 38.105.229.116 s=SIP Media Capabilities c=IN IP4 38.105.229.114 t=0 0 m=audio 14578 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 -------------------------------------------- 2011-06-12 10:50:14.686620 [ERR] sofia.c:5323 Codec Error! v=0 o=ATSI-B2 188 2 IN IP4 208.37.200.235 s=sip call c=IN IP4 208.37.200.245 t=0 0 m=audio 35538 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 -------------------------------------------------------- 2011-06-12 10:47:52.362623 [ERR] sofia.c:5323 Codec Error! v=0 o=VoipSwitch 10828 10828 IN IP4 50.30.35.13 s=VoipSIP i=Audio Session c=IN IP4 50.30.35.13 t=0 0 m=audio 9828 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-1 the problem is that when this error happens FS send the cdr immediately but the call still connected, and the provider still with all billing, because of the we see a billing difference with our provider. The error happens with different provider or customer equipment. Some one know if we can configure of freeswitch in some way that the call hangup. The problem isn't fix the codec error, for us the real problem is that the FS do a cdr but the call still connected. Thank a lot in advice. -- Gustavo Espeche EasyIpCall S.R.L. www.easyipcall.com Bv Mitre 517 24° E Cordoba - Argentina Te: +54 - 351 - 4280633

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Invitation to connect on LinkedIn
by Woody Dickson 12 Jun '11

12 Jun '11

LinkedIn ------------ FreeSWITCH, I'd like to add you to my professional network on LinkedIn. - Woody Woody Dickson Managing Director at Power Telecom San Francisco Bay Area Confirm that you know Woody Dickson https://www.linkedin.com/e/xbphn8-gotu5zpw-6s/isd/3186430560/QOqMJZue/ -- (c) 2011, LinkedIn Corporation

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IVR -> Phrase with a parameter
by Avi Marcus 11 Jun '11

11 Jun '11

I'm kinda new to IVRs and Phrases, but I see this: <action application="phrase" data="msgcount,130"/> <macro name="msgcount"> <input pattern="(.*)"> Meaning I can pass a parameter to the phrase, and then set different matching input configurations. Now, I see phrases are being used in the IVR: I'd like to have an announcement of a balance at the beginning, but start listening for the digits already. But, I'd like to have two different types of announcement in the beginning. I can either duplicate and create two different IVRs and two different sets of phrases, or I can take advantage of that parameter passing. But when I try: greet-long="phrase:cc_ivr_main_menu,${cash}" I get: 2011-06-12 00:05:24.788910 [ERR] switch_ivr_play_say.c:142 Can't find macro cc_ivr_main_menu,33.09. Am I missing something? Do I need to make a lua IVR to start capturing digits while I play other audio (how??) or create split the IVR into two nearly duplicate ones? Thanks, Avi

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Default Install Extensions NEWBIE Question
by Mehma Sarja 11 Jun '11

11 Jun '11

The default install on ubuntu renders a working system servicing 19 extensions. Can I use this as-is in a production environment which only requires within-LAN, inter-departmental voice connectivity? Mehma

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