- FreeSWITCH-users - FreeSWITCH Mailing Lists

More than One DID
by Michael Gende 29 Sep '09

29 Sep '09

Say, could someone please direct me to information on registering more than one DID with a SIP provider? When I try this using two XML files in ~/conf/sip_profiles/external, I find only one or the other registers (both work fine when I use them separately). I don't think the issue is with the provider as we had both DIDs working on our old VoIP server. I'm using the same info. I'm still pretty new, so I suspect the answer is in front of my face. I could use a nudge. Regards, Mike G.

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Transfer oddity
by Bradley Brashier 29 Sep '09

29 Sep '09

Hi guys, I've got a strange situation that I'm at a loss to explain. With all callers, I go through a dialplan where I check to see if they should be a moderator, then transfer them to another which puts them into a conference accordingly. This worked great on one server, but when I copied the code to another server (both running CentOS), the transfer no longer works properly. Here's a log snippet from the incorrectly working server: Dialplan: sofia/internal/14258291243(a)10.10.67.190 Regex (FAIL) [hold_music] destination_number(7001) =~ /^9999$/ break=on-false Dialplan: sofia/internal/14258291243(a)10.10.67.190 Regex (FAIL) [hold_music] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=on-false Dialplan: sofia/internal/14258291243(a)10.10.67.190 ANTI-Action set(zrtp_enrollment=true) Dialplan: sofia/internal/14258291243(a)10.10.67.190 ANTI-Action answer() Dialplan: sofia/internal/14258291243(a)10.10.67.190 ANTI-Action playback(/usr/local/freeswitch/sounds/vpbx/moh.wav) 2009-09-25 07:51:31.204920 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/14258291243(a)10.10.67.190) State Change CS_ROUTING -> CS_EXECUTE 2 (note that the 7001 in the first line is the number I chose for my dialplan) On the working server, the first line is still there, but the second (and further) is replaced by further checks to see if it might be my conference dialplans, which is what I would expect. I looked into dialplans/default.xml, and the code for the above is there, but let me copy it here again to discuss: <extension name="hold_music"> <condition field="destination_number" expression="^9999$"/> <condition field="${sip_has_crypto}" expression="^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$"> <action application="answer"/> <action application="execute_extension" data="is_secure XML features"/> <action application="playback" data="$${hold_music}"/>  <anti-action application="set" data="zrtp_enrollment=true"/> <anti-action application="answer"/> <anti-action application="playback" data="$${hold_music}"/> </condition> </extension> Now, the way I understand this, it says that if the number is 9999, it should check the 2nd condition (which says to play hold music in a couple of different flavors), but if the number is NOT 9999, it should go past, not even checking the 2nd condition. This understanding is corroborated by the "working" server, which does indeed skip past and not check the 2nd condition. Does anyone know why a server might be going into a conditional that it knows it failed on? For what it's worth, both servers are running on the current trunk, with the only change being the addition of flite. BB

1 1

"Proxy|Bypass Media" Wrong Payload.
by Mariano de Llano 29 Sep '09

29 Sep '09

Hi, I'm having a strange behavior with the FS when I'm using it with "inboud-late-negotiation=true" and with the both scenarios "proxy- media=true" or "bypass-media=true". The FS is acting as a pseudo proxy (I know that it is not intend for that). The configuration is similar to this: [Endpoints] <= a => [FS1] <= b => [FS2] Where FS1 is acting as a proxy and registrar. The other one will simply handle the calls between endpoints or with a Gateway. Basically the problem is when I'm trying to call to an AddPac Endpoint, then FS2 sends a call to the FS1 and this one do the proxy or bypass process. Everything works fine with most of the UA (Grandstream/Sipura/Linksys/Xlite/Zoiper/etc) but with the AddPac the FS1 is returning a Payload 96 which seams to be wrong. The only different (In Comparison to the other UA) that I've seen is the "a=ptime:20" in the 200/OK/SDP witch is teorically right. I've seen some posts with a similar issue las year: http://www.mail-archive.com/freeswitch-users@lists.freeswitch.org/msg01468.… Another interesting point is when the media is handle by FS1 ("proxy- media=false"|"bypass-media=false") everything works fine. Also, what is more weird is when the call is generated by the AddPac the SDP everything works fine too. Probably is an AddPac issue, due is the only one failing, however since I have not found something wrong in the SIP capture I'm starting to have my doubts... I'm running Freeswitch 1.0.4pre9. Here is a small flow of the call: 1) FS2 INVITE TO FS1 2) FS1 INVITE TO AddPac Endpoint 3) AddPac Endpoint responds 200 with apparently correct SDP. 4) FS1 responds 200/OK with an Incorrect Payload Here are the corresponding SIP packet trace with the previous call flow: Packet 1 ======= INVITE sip:9499@[FS1 IP] SIP/2.0 Via: SIP/2.0/UDP [FS2 IP]:5060;branch=z9hG4bK60c74dbf;rport From: "[Source Number]" <sip:[Source Number]@[FS2 IP]>;tag=as3bd06925 To: <sip:9499@[FS1 IP]> Contact: <sip:[Source Number]@[FS2 IP]> Call-ID: 14b07c2014d3a0aa7719efa03d979eed@[FS2 IP] CSeq: 102 INVITE User-Agent: Legacy Max-Forwards: 70 Remote-Party-ID: "[Source Number]" <sip:[Source Number]@[FS2 IP]>;privacy=off;screen=no Date: Mon, 28 Sep 2009 15:28:19 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Type: application/sdp Content-Length: 240 v=0 o=root 16330 16330 IN IP4 [FS2 IP] s=session c=IN IP4 [FS2 IP] t=0 0 m=audio 12558 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv Packet 2 ======= INVITE sip:9499@[AddPac IP] SIP/2.0 Via: SIP/2.0/UDP [FS1 IP];rport;branch=z9hG4bKXt9pr4er2H2Np Max-Forwards: 69 From: "[Source Number]" <sip:[Source Number]@[FS1 IP]>;tag=7ZDB1S0pFpSja To: <sip:9499@[AddPac IP]> Call-ID: 765f438b-26e6-122d-3490-51d73cb8d94c CSeq: 120957528 INVITE Contact: <sip:mod_sofia@[FS1 IP]:5060> User-Agent: Proxy 1.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session- description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 228 Remote-Party-ID: <[Source Number]> v=0 o=root 16330 16330 IN IP4 [FS2 IP] s=session c=IN IP4 [FS1 IP] t=0 0 m=audio 27382 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 Packet 3 ======= SIP/2.0 200 OK Via: SIP/2.0/UDP [FS1 IP];rport;branch=z9hG4bKXt9pr4er2H2Np From: "[Source Number]" <sip:[Source Number]@[FS1 IP]>;tag=7ZDB1S0pFpSja To: <sip:9499@[AddPac IP]>;tag=a34a0003a4 Call-ID: 765f438b-26e6-122d-3490-51d73cb8d94c CSeq: 120957528 INVITE Supported: timer, replaces, early-session User-Agent: AddPac SIP Gateway Contact: sip:9499@[AddPac IP] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, UPDATE, REFER, NOTIFY, INFO Content-Type: application/sdp Content-Length: 235 v=0 o=9499 1254144425 1254144425 IN IP4 [AddPac IP] s=AddPac Gateway SDP c=IN IP4 [AddPac IP] t=1254144425 0 m=audio 23004 RTP/AVP 0 101 a=ptime:20 a=rtpmap:0 PCMU/8000/3 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 Packet 4 ======= SIP/2.0 200 OK Via: SIP/2.0/UDP [FS2 IP]:5060;branch=z9hG4bK60c74dbf;rport=5060 From: "[Source Number]" <sip:[Source Number]@[FS2 IP]>;tag=as3bd06925 To: <sip:9499@[FS1 IP]>;tag=6pmjZyFKjD3Ze Call-ID: 14b07c2014d3a0aa7719efa03d979eed@[FS2 IP] CSeq: 102 INVITE Contact: <sip:mod_sofia@[FS1 IP]:5060;transport=udp> User-Agent: Proxy 1.1 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session- description, presence.winfo, message-summary, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 221 v=0 o=9499 1254144425 1254144425 IN IP4 [AddPac IP] s=AddPac Gateway SDP c=IN IP4 [FS1 IP] t=1254144425 0 m=audio 0 RTP/AVP 96 101 a=rtpmap:96 PCMU/8000/3 a=rtpmap:101 telephone-event/8000/1 a=fmtp:101 0-15 ACK sip:mod_sofia@[FS1 IP]:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP [FS2 IP]:5060;branch=z9hG4bK56269b6a;rport From: "[Source Number]" <sip:[Source Number]@[FS2 IP]>;tag=as3bd06925 To: <sip:9499@[FS1 IP]>;tag=6pmjZyFKjD3Ze Contact: <sip:[Source Number]@[FS2 IP]> Call-ID: 14b07c2014d3a0aa7719efa03d979eed@[FS2 IP] CSeq: 102 ACK User-Agent: Legacy Max-Forwards: 70 Remote-Party-ID: "[Source Number]" <sip:[Source Number]@[FS2 IP]>;privacy=off;screen=no Content-Length: 0 I've tried the same test with different codecs, and it is always messing with the payload. Thanks in advance Regards M

2 5

Dynamic volume adjustment in heterogenous IVR menus
by Alberto Escudero 29 Sep '09

29 Sep '09

Sorry for the email subject that sounds like a IEEE paper. I am building IVRs using FS API and sending out audio that is a combination of TTS and playing WAV files. What is the best way to control volume levels? I know i might be asking for magic here... In any case, is there any simple ways to add gain to certain nodes of am IVR? /aep -- Stopping junk mailers is good for the environment

2 1

Delay when transferring call
by Sias Mey 29 Sep '09

29 Sep '09

Hi, Im having a strange issue with a api triggered call transfer. There seems to be a long delay between when the transfer is triggered and when it actually happens. 2009-09-17 11:36:26.995001 [NOTICE] switch_ivr.c:1350 Transfer sofia/internal/1004(a)192.168.0.10 to xml [incust-camp=lucidlive-call=78-conf=41@default] Error in my_thread_global_end(): 26 threads didn't exit 2009-09-17 11:36:31.997191 [INFO] mod_dialplan_xml.c:315 Processing 1004->incust-camp=lucidlive-call=7 8-conf=41 in context default 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Execution start 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Connecting to Ringback to add call 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Finished adding calls 2009-09-17 11:36:31.997191 [INFO] regin.js:1 Connecting to database lucidlive to update call 2009-09-17 11:36:32.37433 [INFO] regin.js:1 Finished updateing call I though it was my regin.js script causing the delay since it runs a couple of database queries and other things, but the output above show that runs fine. My question is about the delay between 11:36:26 -> 11:36:31. The call is being transfered out of a fifo, but for those 5 seconds there is no MOH or anything else. Just silence. The transfer is triggered via a xml rpc call. But since the delay is between the switch_ivr and mod_dialplan_xml somewhere I doubt that that has much to do with it. Any clues or other places I can go look?. Cheers, Sias

2 3

SILK speech codec, Celt, FreeSWITCH
by Jason White 29 Sep '09

29 Sep '09

I just read a blog post about this. http://tools.ietf.org/html/draft-vos-silk-00 My main questions about it would be 1. whether it will be licenced in such a way as to permit free software/open-source implementation in FreeSWITCH. I suppose this partly depends on IETF's patent policy. 2. How it compares with Celt, which doesn't give rise to any known patent issues and is available for FreeSWITCH today. for other FreeSWITCH users who wish to experiment with Celt and listen to high-quality wide-band calls, it should be noted that Celt is still under development and backward-incompatible changes have occurred in the past. Making sure that both endpoints are using the same version of Celt solves this, otherwise the audible results can be rather unpleasant.

4 7

confuse of regex command with "|"
by Seven Du 29 Sep '09

29 Sep '09

Hi, is this a bug? freeswitch@internal> regex "10"|"09|10" false freeswitch@internal> regex "10"|"10" true freeswitch@internal> regex "10"|"(09|10)" false freeswitch@internal> 2009-09-27 11:47:00.815355 [ERR] switch_regex.c:101 COMPILE ERROR: 4 [missing )]["(09] the first one should be true?

3 3

Load-Balance FreeSWITCH with Ultramonkey
by Mike van Lammeren 28 Sep '09

28 Sep '09

Hello! I followed a tutorial that describes load-balancing Asterisk with Ultramonkey, but cannot get it to work with FreeSWITCH: http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf My X-Lite client fails to register with the server. I have looked at the packets with wireshark, and found that when X-Lite sends a SIP Register packet, it gets an ICMP response: Destination Unreachable (Port Unreachable). Has anyone got a load-balanced FreeSWITCH setup working? Mike van Lammeren

4 3

No ring tone while recording incoming call. Please help.
by Svetik VOIP 28 Sep '09

28 Sep '09

Hi guys, I have solve my problem by adding <action application="ring_ready" /> before <action application="set" data="ringback=${us-ring}"/> I was looking at the http://wiki.freeswitch.org/wiki/Home_PBX_Example and noticed this line. Tried it and it works like a charm. Thanks everybody, especially Brian and MC. Igor > Brian, >> >> Thank you very much for your reply. >> >> I have tried to add transfer_ringback action, but it did not solve my >> problem. >> Destination phone is ringing, but the person who is calling does not >> hear ringing tone in hte handset. >> >> Is there anything in the logfile that can help you to identify the problem? >> >What kind of system is the calling party connected to? It looks like a >180 is sent out by FS: > >2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel sofia/internal/ >sip:main at 192.168.0.121:5060 entering state [proceeding][180] > >At that point the server at the originating side *should* generate >pretend ringing for the calling phone. If that is not happening then >you need to see what's going on at the originating side. Is it a SIP provider? > >-MC > > >> >> Closest I can see is: >> 2009-09-22 17:18:05.444402 [DEBUG] switch_ivr_originate.c:1738 Raw >> Codec Activation Success L16 at 8000hz 1 channel 20ms >> 2009-09-22 17:18:05.444402 [DEBUG] switch_ivr_originate.c:1797 Play >> Ringback Tone [%(2000,4000,440.0,480.0)] >> 2009-09-22 17:18:05.447237 [DEBUG] switch_core_io.c:232 >> sofia/external/ >> 4163641113 at 67.205.74.164 receive message [TRANSCODING_NECESSARY] >> 2009-09-22 17:18:05.463192 [DEBUG] sofia.c:3312 Channel >> sofia/internal/ sip:main at 192.168.0.121:5060 entering state >> [proceeding][180] >> 2009-09-22 17:18:05.463192 [NOTICE] sofia.c:3376 Ring-Ready >> sofia/internal/ sip:main at 192.168.0.121:5060! >> 2009-09-22 17:18:14.739182 [DEBUG] sofia.c:3312 Channel >> sofia/external/ >> 4163641113 at 67.205.74.164 entering state [terminated][487] >> 2009-09-22 17:18:14.739182 [NOTICE] sofia.c:3873 Hangup >> sofia/external/ >> 4163641113 at 67.205.74.164 [CS_EXECUTE] [ORIGINATOR_CANCEL] >> >> Thank you, >> >> Igor >> >> >set ringback before record_session and also set transfer_ringback >> >because record_session causes an pre-answer. >> > >> >/b >> > >> >On Sep 21, 2009, at 2:13 PM, Svetik VOIP wrote: >> > >> >> Hi, >> >> >> >> I have trouble recording incoming calls with FreeSwitch. >> >> >> >> I have followed the instruction from Misc. Dialplan Tools record >> >> session >> >> (http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_sessi >> >> on) It works well for outgoing calls, but I have the problem with >> >> incoming calls. >> >> >> >> The person who is calling does not hear ring tone, he hears just >> >> the silence until I pick up the phone. Everything else is working, >> >> we can talk, conversation is recorded. >> >> >> >> Here is a copy of my dialplan for incoming calls >> >> /usr/local/freeswitch/conf/dialplan/public/voipms.xml >> >> >> >> <include> >> >> <extension name="voipms">  >> >> <condition field="destination_number" >> >> expression="XXXXXXXXXX">  >> >> <action application="set" data="RECORD_TITLE=Recording >> >> $ {destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H: >> >> %M)}"/> >> >> <action application="set" data="RECORD_COPYRIGHT=(c) >> >> 2009"/> >> >> <action application="set" >> >> data="RECORD_SOFTWARE=FreeSwitch"/> >> >> <action application="set" >> >> data="RECORD_ARTIST=FreeSwitch"/> >> >> <action application="set" >> >> data="RECORD_COMMENT=FreeSwitch"/> >> >> <action application="set" data="RECORD_DATE=${strftime >> >> (%Y-%m-%d %H:%M)}"/> >> >> <action application="set" data="RECORD_STEREO=true"/> >> >> <action application="set" data="RECORD_ANSWER_REQ=true"/> >> >> <action application="set" data="ringback=${us-ring}"/> >> >> <action application="record_session" >> >> data="$${base_dir}/ >> >> recordings/archive/${strftime(%Y-%m-%d-%H-%M-%S)}_$ >> >> {destination_number}_${caller_id_number}.wav"/> >> >> <action application="bridge" data="user/user1@$ >> >> {domain_name}"/> >> >> </condition> >> >> </include> >> > >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> http://www.freeswitch.org >> >

3 3

SUBSCRIBE and NOTIFY
by Rudá Cunha 28 Sep '09

28 Sep '09

I'm having to configure FreeSWITCH. Baxei version 1.0.4 and I am accessing with the users 1000 and 1001. I register, make the connection. But I'm trying to see to see who is connected (SUBSCRIBE and NOTIFY). But sometimes you work, sometimes you do I connect the other User does not receive the information that I connected (I sent the (SUBSCRIBE and NOTIFY)). That is, sometimes you and I'm connected to another user I'm not, sometimes you work.

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