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playing value with mod_say & getting DTMF input at the same time
by Mi Ke 09 Aug '11

09 Aug '11

Hi All, Using mod_say and read function at once - is that possible with Lua ? I need to play a numeric value (like credit amount or time) to Leg A and catch user's input at the same time. As a temporary solution, I wrote a function translating numeric values into a string "prompt1.wav&prompt2.wav....etc" and then suppliy is to the read function. Is there anyway I could use say for that ? Thanks / Mike

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Problem with XML CDR
by Nice Voip 09 Aug '11

09 Aug '11

Dear All, This problem is very hard to reproduce and i really don't know when it would happen, i don't have sip log or other traces but only the CDR file and its looks like this: <caller_id_name>TELESAT TRAJKOVIÃ <ani>+3xxxxxxx</ani> ? MILE</caller_id_name> <aniii></aniii> and sometimes <oritinate_early_media> is writted <originate_uarly_media> (insead of early its "uarly") i noted down this issue on: FreeSWITCH Version 1.0.head (git-1d3417a 2011-06-07 17-35-49 -0400) Thanks.

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SIP MESSAGE delivery via Freeswitch
by Soomro, Amjad 09 Aug '11

09 Aug '11

Hello, I am using PJSUA to send SIP MESSAGE to another SIP URL. The message is received at FreeSwitch server and 200 OK response is sent back to the sender. However, I see no message being forwarded to the receiver at the server. Both wireshark and siptrace reveal no attempt to deliver the message to the receipient. The SIP message and its response is: ------------------------------------------------------------------------ recv 572 bytes from udp/[96.57.214.157]:1024 at 16:16:19.536837: ------------------------------------------------------------------------ MESSAGE sip:1001@wimed.philips.com SIP/2.0 Via: SIP/2.0/UDP 96.57.214.157:1024;rport;branch=z9hG4bKPjf03b25a7-3d05-46b1-bc16-347cbbd97c07 Max-Forwards: 70 From: <sip:1000@wimed.philips.com>;tag=9ef1d5f6-d19c-459c-87cf-eaab7ca9dfe2 To: <sip:1001@wimed.philips.com> Call-ID: 907e1671-1b1f-495e-9808-5802e7670328 CSeq: 29555 MESSAGE Accept: text/plain, application/im-iscomposing+xml Contact: <sip:1000@96.57.214.157:1024;transport=UDP;ob> User-Agent: PJSUA v1.10.0 Linux-2.6.35.13/i686/glibc-2.13 Content-Type: text/plain Content-Length: 16 SIP REGISTRATION ------------------------------------------------------------------------ send 541 bytes to udp/[96.57.214.157]:1024 at 16:16:19.537837: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 96.57.214.157:1024;rport=1024;branch=z9hG4bKPjf03b25a7-3d05-46b1-bc16-347cbbd97c07 From: <sip:1000@wimed.philips.com>;tag=9ef1d5f6-d19c-459c-87cf-eaab7ca9dfe2 To: <sip:1001@wimed.philips.com>;tag=00tQ14HpaSDUg Call-ID: 907e1671-1b1f-495e-9808-5802e7670328 CSeq: 29555 MESSAGE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ---------------------------------------------------------------- The registration messages from both the clients at Freeswitch are: ------------------------------------------------------------------------ recv 841 bytes from udp/[96.57.214.157]:1024 at 16:15:58.158614: ------------------------------------------------------------------------ REGISTER sip:96.57.214.156 SIP/2.0 Via: SIP/2.0/UDP 96.57.214.157:1024;rport;branch=z9hG4bKPj4b9b314a-e7d7-4e51-9b58-2aab22e78633 Max-Forwards: 70 From: <sip:1000@wimed.philips.com>;tag=738d0018-6309-47df-81a8-0efbcb12a1b7 To: <sip:1000@wimed.philips.com> Call-ID: 5887dcaf-c8b5-4837-a708-ea8c75704f79 CSeq: 14535 REGISTER User-Agent: PJSUA v1.10.0 Linux-2.6.35.13/i686/glibc-2.13 Contact: <sip:1000@96.57.214.157:1024;transport=UDP;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="1000", realm="wimed.philips.com", nonce="bba2277d-66b8-4d56-93bf-df343dc7fe52", uri="sip:96.57.214.156", response="b2a31bef17c12d8c0c6c3961ae958932", algorithm=MD5, cnonce="eb828e16-22b4-4a24-ac21-9b96bc0ad765", qop=auth, nc=00000001 Content-Length: 0 ------------------------------------------------------------------------ send 647 bytes to udp/[96.57.214.157]:1024 at 16:15:58.185616: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 96.57.214.157:1024;rport=1024;branch=z9hG4bKPj4b9b314a-e7d7-4e51-9b58-2aab22e78633 From: <sip:1000@wimed.philips.com>;tag=738d0018-6309-47df-81a8-0efbcb12a1b7 To: <sip:1000@wimed.philips.com>;tag=X5eDvKZBKya3D Call-ID: 5887dcaf-c8b5-4837-a708-ea8c75704f79 CSeq: 14535 REGISTER Contact: <sip:1000@96.57.214.157:1024;transport=UDP;ob>;expires=30 Date: Tue, 09 Aug 2011 16:15:58 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ ------------------------------------------------------------------------ recv 841 bytes from udp/[96.57.214.157]:5060 at 16:15:56.372512: ------------------------------------------------------------------------ REGISTER sip:96.57.214.156 SIP/2.0 Via: SIP/2.0/UDP 96.57.214.157:5060;rport;branch=z9hG4bKPj53492c33-27ad-42a5-a5ea-6b0a7fb6760c Max-Forwards: 70 From: <sip:1001@wimed.philips.com>;tag=76e9c9d5-7f52-4f22-a17c-11a8c287c227 To: <sip:1001@wimed.philips.com> Call-ID: 71251231-581e-4664-85b9-a9cd74877ca5 CSeq: 52991 REGISTER User-Agent: PJSUA v1.10.0 Linux-2.6.35.13/i686/glibc-2.13 Contact: <sip:1001@96.57.214.157:5060;transport=UDP;ob> Expires: 300 Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Authorization: Digest username="1001", realm="wimed.philips.com", nonce="aac8bc16-1644-4986-a4e2-a5749d1ea911", uri="sip:96.57.214.156", response="a5a19227c03a4d053ccb0dd65af23aaf", algorithm=MD5, cnonce="33e8968a-5f8f-45d3-9ccf-a3e20b50de66", qop=auth, nc=00000001 Content-Length: 0 ------------------------------------------------------------------------ send 647 bytes to udp/[96.57.214.157]:5060 at 16:15:56.396514: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 96.57.214.157:5060;rport=5060;branch=z9hG4bKPj53492c33-27ad-42a5-a5ea-6b0a7fb6760c From: <sip:1001@wimed.philips.com>;tag=76e9c9d5-7f52-4f22-a17c-11a8c287c227 To: <sip:1001@wimed.philips.com>;tag=ta32p2c1U37aB Call-ID: 71251231-581e-4664-85b9-a9cd74877ca5 CSeq: 52991 REGISTER Contact: <sip:1001@96.57.214.157:5060;transport=UDP;ob>;expires=30 Date: Tue, 09 Aug 2011 16:15:56 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.head-git- Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------ Are there any additional settings (other than default) at Freeswitch server for messages to be actually forwarded to the receipient? Thanks in advance Amjad ________________________________ The information contained in this message may be confidential and legally protected under applicable law. The message is intended solely for the addressee(s). If you are not the intended recipient, you are hereby notified that any use, forwarding, dissemination, or reproduction of this message is strictly prohibited and may be unlawful. If you are not the intended recipient, please contact the sender by return e-mail and destroy all copies of the original message.

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how to have fs interrupt a file or tts if digit is pressed
by covici＠ccs.covici.com 09 Aug '11

09 Aug '11

Hi. I am using Perl, if that makes any difference and I would like to arrange things in such a way that during a prompt, if the person presses a key, then the prompt will stop speaking and I can see what the key is and do something. Here is an excerpt of the script I am using. $session->setInputCallback('got_press',""); #listen for key presses in the background while($session->ready()) { $session->streamFile("test_break.wav"); if ($press_so_far != "") { $session->say($press_so_far,"EN", "NAME_SPELLED", "ITERATED"); $press_so_far=""; } } Instead of streamFile, I tried speaking text through tts_command line, and a phrase macro which did the same. I was sure that at least streamFile would break, but it did not. Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici(a)ccs.covici.com

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Core dump - When using user_recurse_variables in bridge
by Nagalenoj H. 09 Aug '11

09 Aug '11

Hi Friends, I've got segfault in yesterday's git, when I execute the below from nc, sendmsg call-command: execute execute-app-name: bridge execute-app-arg: {user_recurse_variables=false,group_confirm_cancel_timeout=true,group_confirm_key=exec,group_confirm_file='perl /FMS-FS/bin/bridge_confirm.pl'}[leg_timeout=10]user/1001 It works fine when I bridge without any of those variables. execute-app-arg: [leg_timeout=10]user/1001 Here is the backtrace, http://pastebin.freeswitch.org/17004. -- Regards, Nagalenoj H.

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intermittent EXCHANGE_ROUTING_ERROR
by Stephen Wilkey 09 Aug '11

09 Aug '11

Thanks for the discussion on this issue. I have discovered that my ISP is sending calls to me with Max-Forwards=4 which means that I can't do much with the call when it arrives. Is there a way to ignore or override the Max-Forwards setting that is provided with the call when it arrives so that I am able to process it? Regards, Stephen

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rfc 5168 (picture fast update ) support in freeswitch.
by Narendra Sirugudi 09 Aug '11

09 Aug '11

Hi, I wanted to know whether freeswitch/sofia stack has support for rfc 5168 (picture fast update) ? It looks like there is something related picture fast update in mod_sofia. Is there any way to initiate a "picture fast update" SIP INFO request in freeswitch ? Maybe through some application. thanks, --naren

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Look n8nrcuv
by Max Alex 09 Aug '11

09 Aug '11

2qhealspczpb. http://aqabpsychology.co.uk/wp-content/plugins/wp-filemanager/incl/img/yptm… cevvqi eerhlt5es17 57ty43lzfqn, q5kbmgldfmln tavkldcqk8. ziy19sq30 9l2c8hm i9ueccmem3. -- Thanks, Max Alex Voip Developer

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Re: [Freeswitch-users] intermittent EXCHANGE_ROUTING_ERROR
by steve2010 09 Aug '11

09 Aug '11

Yes, please post. Date: Mon, 8 Aug 2011 19:34:44 -0700 From: ml-node+6666783-1527079717-350115(a)n2.nabble.com To: stephen_wilkey(a)hotmail.com Subject: Re: intermittent EXCHANGE_ROUTING_ERROR Thanks for the discussion on this issue. I have discovered that my ISP is sending calls to me with Max-Forwards=4 which means that I can't do much with the call when it arrives. Is there a way to ignore or override the Max-Forwards setting that is provided with the call when it arrives so that I am able to process it? Regards, Stephen Bill W.-2-3 wrote: Hey Anthony/Michael, After a lot of digging, it turns out the problem was the Max-Forwards on the incoming SIP traffic was too low for FreeSWITCH to complete the call. Thanks so much for all your help! If you reply to this email, your message will be added to the discussion below:http://freeswitch-users.2379917.n2.nabble.com/intermittent-EXCHANGE-R… To unsubscribe from intermittent EXCHANGE_ROUTING_ERROR, click here. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/intermittent-EXCHANGE-ROUTING… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Fidelio
by Nandy Dagondon 09 Aug '11

09 Aug '11

hi everybody, anyone working on interfacing FS with Fidelio Hotel PMS? i can't find the FIAS protocol/specs online. is this freely available? tks, nandy

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