Re: [asterisk-users] Playback with noanswer in AGI
Hi Sammy, Thanks for input. I have an eyebeam softphone registered with Asterisk 1.8.6 locally and from agi, I pass this $filetoplay = 'congestion'; $agi-exec(Progress); $agi-exec(Playback $filetoplay,noanswer); Have tried putting file in .gsm and .wav formats, I hear ringing tone instead of playback Please have a look at sip-trace --- SIP read from UDP:176.249.0.50:8721 --- INVITE sip:100@176.249.0.77 SIP/2.0 To: sip:100@176.249.0.77 From: Zohairsip:1000@176.249.0.77;tag=7f222672 Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport Call-ID: 2932f90ef302332b CSeq: 2 INVITE Contact: sip:1000@176.249.0.50:8721 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3006o stamp 17551 Authorization: Digest username=1000,realm=asterisk,nonce=2abce759,uri=sip:100@176.249.0.77 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5 Content-Length: 269 v=0 o=- 4333518 4333604 IN IP4 176.249.0.50 s=eyeBeam c=IN IP4 176.249.0.50 t=0 0 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 11 lines) --- Sending to 176.249.0.50:8721 (no NAT) sing INVITE request as basis request - 2932f90ef302332b Found peer '1000' for '1000' from 176.249.0.50:8721 == Using SIP RTP CoS mark 5 Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 101 Found audio description format speex for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 176.249.0.50:6506 Looking for 100 in default (domain 176.249.0.77) list_route: hop: sip:1000@176.249.0.50:8721 --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Length: 0 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID) -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php -- AGI Script Executing Application: (Progress) Options: () Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77;tag=as01491743 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Type: application/sdp Content-Length: 258 v=0 o=root 1225456982 1225456982 IN IP4 176.249.0.77 s=Asterisk PBX 1.8.0 c=IN IP4 176.249.0.77 t=0 0 m=audio 15918 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- AGI Script Executing Application: (Playback) Options: (congestion,noanswer) -- SIP/1000-0019 Playing 'congestion.slin' (language 'en') -- SIP/1000-0019AGI Script agi.php completed, returning 0 Regards, Zohair Raza On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote: Hey Danny, I've this thing exactly running and working as Zohair mentioned! i.e I do not answer() the call rather put a progress() and soon after that playing back the sound file from playback with noanswer and then I get the file streaming as 183-Session progress file. I do understand that playing any sound file before establishing any audio session between two end point will result in no-adio from playback() BUT the combination of progress() and playback(,noanswer) works fine for me. What I think the issue could be for Zohair is that its requesting/incoming session(carrier) isn't allowing the 183-Session progress. Zohair can you do a SIP trace for this particular call along with the dialplan executing for it!? Regards, Sammy. On Tue, Feb 7, 2012 at 11:55 AM, Zohair Raza engineerzuhairr...@gmail.com wrote:
Re: [asterisk-users] Playback with noanswer in AGI
Sammy, Problem is at phones, with a linksys phone it works but with eyebeam and fanvill it doesn't Maybe they don't support early media. I think i will have to stick with ResetCDR and that will be okay now as I've modified the code for that Thank you Regards, Zohair Raza On Tue, Feb 7, 2012 at 12:09 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: Hi Sammy, Thanks for input. I have an eyebeam softphone registered with Asterisk 1.8.6 locally and from agi, I pass this $filetoplay = 'congestion'; $agi-exec(Progress); $agi-exec(Playback $filetoplay,noanswer); Have tried putting file in .gsm and .wav formats, I hear ringing tone instead of playback Please have a look at sip-trace --- SIP read from UDP:176.249.0.50:8721 --- INVITE sip:100@176.249.0.77 SIP/2.0 To: sip:100@176.249.0.77 From: Zohairsip:1000@176.249.0.77;tag=7f222672 Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport Call-ID: 2932f90ef302332b CSeq: 2 INVITE Contact: sip:1000@176.249.0.50:8721 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3006o stamp 17551 Authorization: Digest username=1000,realm=asterisk,nonce=2abce759,uri= sip:100@176.249.0.77 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5 Content-Length: 269 v=0 o=- 4333518 4333604 IN IP4 176.249.0.50 s=eyeBeam c=IN IP4 176.249.0.50 t=0 0 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 11 lines) --- Sending to 176.249.0.50:8721 (no NAT) sing INVITE request as basis request - 2932f90ef302332b Found peer '1000' for '1000' from 176.249.0.50:8721 == Using SIP RTP CoS mark 5 Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 101 Found audio description format speex for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 176.249.0.50:6506 Looking for 100 in default (domain 176.249.0.77) list_route: hop: sip:1000@176.249.0.50:8721 --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Length: 0 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID) -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php -- AGI Script Executing Application: (Progress) Options: () Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77;tag=as01491743 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Type: application/sdp Content-Length: 258 v=0 o=root 1225456982 1225456982 IN IP4 176.249.0.77 s=Asterisk PBX 1.8.0 c=IN IP4 176.249.0.77 t=0 0 m=audio 15918 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- AGI Script Executing Application: (Playback) Options: (congestion,noanswer) -- SIP/1000-0019 Playing 'congestion.slin' (language 'en') -- SIP/1000-0019AGI Script agi.php completed, returning 0 Regards, Zohair Raza On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote: Hey Danny, I've this thing exactly running and working as Zohair mentioned! i.e I do not answer() the call rather put a progress() and soon after that playing back the sound file from playback with noanswer and then I get the file streaming as 183-Session progress file. I do understand that playing any sound file before establishing any audio session
Re: [asterisk-users] MixMonitor and ChanSpy
On 02/02/2012 11:24 AM, Jonas Kellens wrote: Hello, ChanSpy can not be used on a Channel that is being recorded with MixMonitor. How can I verify if a channel which I want to spy on, is currently not being recorded ?! Anyone with some feedback ?! I notice that ongoing recordings are temporarily saved in the directory /tmp. How could I look from the dialplan into the /tmp-directory to see if there is an ongoing recording for the channel that one wants to spy on ? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Playback with noanswer in AGI
Hi, Given invites seems fine, can you take a wireshark trace of the call on your eyebeam machine! from that wireshark trace use telephony calls options and hear if you are actually receiving RTPs on your system. If you could hear the played back sound file on your eyembeam machine . this would mean that your eyebeam client is not good enough to play media while its in 183 session progress. Also can you send me the short sample php-agi script you are executing so i actually test this on my virtual machines as well. Regards, Sammy On Tue, Feb 7, 2012 at 1:09 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: Hi Sammy, Thanks for input. I have an eyebeam softphone registered with Asterisk 1.8.6 locally and from agi, I pass this $filetoplay = 'congestion'; $agi-exec(Progress); $agi-exec(Playback $filetoplay,noanswer); Have tried putting file in .gsm and .wav formats, I hear ringing tone instead of playback Please have a look at sip-trace --- SIP read from UDP:176.249.0.50:8721 --- INVITE sip:100@176.249.0.77 SIP/2.0 To: sip:100@176.249.0.77 From: Zohairsip:1000@176.249.0.77;tag=7f222672 Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport Call-ID: 2932f90ef302332b CSeq: 2 INVITE Contact: sip:1000@176.249.0.50:8721 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3006o stamp 17551 Authorization: Digest username=1000,realm=asterisk,nonce=2abce759,uri= sip:100@176.249.0.77 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5 Content-Length: 269 v=0 o=- 4333518 4333604 IN IP4 176.249.0.50 s=eyeBeam c=IN IP4 176.249.0.50 t=0 0 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 11 lines) --- Sending to 176.249.0.50:8721 (no NAT) sing INVITE request as basis request - 2932f90ef302332b Found peer '1000' for '1000' from 176.249.0.50:8721 == Using SIP RTP CoS mark 5 Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 101 Found audio description format speex for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 176.249.0.50:6506 Looking for 100 in default (domain 176.249.0.77) list_route: hop: sip:1000@176.249.0.50:8721 --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Length: 0 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID) -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php -- AGI Script Executing Application: (Progress) Options: () Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77;tag=as01491743 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Type: application/sdp Content-Length: 258 v=0 o=root 1225456982 1225456982 IN IP4 176.249.0.77 s=Asterisk PBX 1.8.0 c=IN IP4 176.249.0.77 t=0 0 m=audio 15918 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- AGI Script Executing Application: (Playback) Options: (congestion,noanswer) -- SIP/1000-0019 Playing 'congestion.slin' (language 'en') -- SIP/1000-0019AGI Script agi.php completed, returning 0 Regards, Zohair Raza On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote: Hey Danny, I've this thing exactly running and working as Zohair mentioned! i.e I do not answer() the
Re: [asterisk-users] Playback with noanswer in AGI
Exactly that's what I expected. Great - now have fun On Tue, Feb 7, 2012 at 2:09 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: Sammy, Problem is at phones, with a linksys phone it works but with eyebeam and fanvill it doesn't Maybe they don't support early media. I think i will have to stick with ResetCDR and that will be okay now as I've modified the code for that Thank you Regards, Zohair Raza On Tue, Feb 7, 2012 at 12:09 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hi Sammy, Thanks for input. I have an eyebeam softphone registered with Asterisk 1.8.6 locally and from agi, I pass this $filetoplay = 'congestion'; $agi-exec(Progress); $agi-exec(Playback $filetoplay,noanswer); Have tried putting file in .gsm and .wav formats, I hear ringing tone instead of playback Please have a look at sip-trace --- SIP read from UDP:176.249.0.50:8721 --- INVITE sip:100@176.249.0.77 SIP/2.0 To: sip:100@176.249.0.77 From: Zohairsip:1000@176.249.0.77;tag=7f222672 Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport Call-ID: 2932f90ef302332b CSeq: 2 INVITE Contact: sip:1000@176.249.0.50:8721 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3006o stamp 17551 Authorization: Digest username=1000,realm=asterisk,nonce=2abce759,uri= sip:100@176.249.0.77 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5 Content-Length: 269 v=0 o=- 4333518 4333604 IN IP4 176.249.0.50 s=eyeBeam c=IN IP4 176.249.0.50 t=0 0 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 11 lines) --- Sending to 176.249.0.50:8721 (no NAT) sing INVITE request as basis request - 2932f90ef302332b Found peer '1000' for '1000' from 176.249.0.50:8721 == Using SIP RTP CoS mark 5 Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 101 Found audio description format speex for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 176.249.0.50:6506 Looking for 100 in default (domain 176.249.0.77) list_route: hop: sip:1000@176.249.0.50:8721 --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Length: 0 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID) -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php -- AGI Script Executing Application: (Progress) Options: () Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77;tag=as01491743 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Type: application/sdp Content-Length: 258 v=0 o=root 1225456982 1225456982 IN IP4 176.249.0.77 s=Asterisk PBX 1.8.0 c=IN IP4 176.249.0.77 t=0 0 m=audio 15918 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- AGI Script Executing Application: (Playback) Options: (congestion,noanswer) -- SIP/1000-0019 Playing 'congestion.slin' (language 'en') -- SIP/1000-0019AGI Script agi.php completed, returning 0 Regards, Zohair Raza On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.com wrote: Hey Danny, I've this thing exactly running and working as Zohair mentioned! i.e I do not answer() the call rather put a progress() and soon after that playing back the sound file from playback with noanswer and then
Re: [asterisk-users] Playback with noanswer in AGI
Yes, Thanks Regards, Zohair Raza On Tue, Feb 7, 2012 at 1:37 PM, Sammy Govind govoi...@gmail.com wrote: Exactly that's what I expected. Great - now have fun On Tue, Feb 7, 2012 at 2:09 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: Sammy, Problem is at phones, with a linksys phone it works but with eyebeam and fanvill it doesn't Maybe they don't support early media. I think i will have to stick with ResetCDR and that will be okay now as I've modified the code for that Thank you Regards, Zohair Raza On Tue, Feb 7, 2012 at 12:09 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hi Sammy, Thanks for input. I have an eyebeam softphone registered with Asterisk 1.8.6 locally and from agi, I pass this $filetoplay = 'congestion'; $agi-exec(Progress); $agi-exec(Playback $filetoplay,noanswer); Have tried putting file in .gsm and .wav formats, I hear ringing tone instead of playback Please have a look at sip-trace --- SIP read from UDP:176.249.0.50:8721 --- INVITE sip:100@176.249.0.77 SIP/2.0 To: sip:100@176.249.0.77 From: Zohairsip:1000@176.249.0.77;tag=7f222672 Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport Call-ID: 2932f90ef302332b CSeq: 2 INVITE Contact: sip:1000@176.249.0.50:8721 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3006o stamp 17551 Authorization: Digest username=1000,realm=asterisk,nonce=2abce759,uri= sip:100@176.249.0.77 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5 Content-Length: 269 v=0 o=- 4333518 4333604 IN IP4 176.249.0.50 s=eyeBeam c=IN IP4 176.249.0.50 t=0 0 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 11 lines) --- Sending to 176.249.0.50:8721 (no NAT) sing INVITE request as basis request - 2932f90ef302332b Found peer '1000' for '1000' from 176.249.0.50:8721 == Using SIP RTP CoS mark 5 Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 101 Found audio description format speex for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 176.249.0.50:6506 Looking for 100 in default (domain 176.249.0.77) list_route: hop: sip:1000@176.249.0.50:8721 --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Length: 0 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID) -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php -- AGI Script Executing Application: (Progress) Options: () Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77;tag=as01491743 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Type: application/sdp Content-Length: 258 v=0 o=root 1225456982 1225456982 IN IP4 176.249.0.77 s=Asterisk PBX 1.8.0 c=IN IP4 176.249.0.77 t=0 0 m=audio 15918 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- AGI Script Executing Application: (Playback) Options: (congestion,noanswer) -- SIP/1000-0019 Playing 'congestion.slin' (language 'en') -- SIP/1000-0019AGI Script agi.php completed, returning 0 Regards, Zohair Raza On Tue, Feb 7, 2012 at 11:38 AM, Sammy Govind govoi...@gmail.comwrote: Hey Danny, I've this thing exactly running and working as Zohair mentioned! i.e I do not answer() the
Re: [asterisk-users] Playback with noanswer in AGI
Confirmed as well, played back with wireshark and audio was there but phone was ringing. Thanks again. Regards, Zohair Raza On Tue, Feb 7, 2012 at 1:37 PM, Sammy Govind govoi...@gmail.com wrote: Hi, Given invites seems fine, can you take a wireshark trace of the call on your eyebeam machine! from that wireshark trace use telephony calls options and hear if you are actually receiving RTPs on your system. If you could hear the played back sound file on your eyembeam machine . this would mean that your eyebeam client is not good enough to play media while its in 183 session progress. Also can you send me the short sample php-agi script you are executing so i actually test this on my virtual machines as well. Regards, Sammy On Tue, Feb 7, 2012 at 1:09 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: Hi Sammy, Thanks for input. I have an eyebeam softphone registered with Asterisk 1.8.6 locally and from agi, I pass this $filetoplay = 'congestion'; $agi-exec(Progress); $agi-exec(Playback $filetoplay,noanswer); Have tried putting file in .gsm and .wav formats, I hear ringing tone instead of playback Please have a look at sip-trace --- SIP read from UDP:176.249.0.50:8721 --- INVITE sip:100@176.249.0.77 SIP/2.0 To: sip:100@176.249.0.77 From: Zohairsip:1000@176.249.0.77;tag=7f222672 Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;rport Call-ID: 2932f90ef302332b CSeq: 2 INVITE Contact: sip:1000@176.249.0.50:8721 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3006o stamp 17551 Authorization: Digest username=1000,realm=asterisk,nonce=2abce759,uri= sip:100@176.249.0.77 ,response=c1a2dbcf1b51d839521b1ee848bea055,algorithm=MD5 Content-Length: 269 v=0 o=- 4333518 4333604 IN IP4 176.249.0.50 s=eyeBeam c=IN IP4 176.249.0.50 t=0 0 m=audio 6506 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 119610F1 00B3 176.249.0.50 6506 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv - --- (13 headers 11 lines) --- Sending to 176.249.0.50:8721 (no NAT) sing INVITE request as basis request - 2932f90ef302332b Found peer '1000' for '1000' from 176.249.0.50:8721 == Using SIP RTP CoS mark 5 Found RTP audio format 100 Found RTP audio format 6 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 5 Found RTP audio format 101 Found audio description format speex for ID 100 Found audio description format telephone-event for ID 101 Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x2012e (gsm|ulaw|alaw|adpcm|g729|speex16)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 176.249.0.50:6506 Looking for 100 in default (domain 176.249.0.77) list_route: hop: sip:1000@176.249.0.50:8721 --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 100 Trying Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Length: 0 -- Executing [100@default:1] AGI(SIP/1000-0019, agi.php,DID) -- Launched AGI Script /var/lib/asterisk/agi-bin/agi.php -- AGI Script Executing Application: (Progress) Options: () Audio is at 5060 Adding codec 0x4 (ulaw) to SDP Adding codec 0x8 (alaw) to SDP Adding non-codec 0x1 (telephone-event) to SDP --- Transmitting (no NAT) to 176.249.0.50:8721 --- SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 176.249.0.50:8721 ;branch=z9hG4bK-d87543-521938753-1--d87543-;received=176.249.0.50;rport=8721 From: Zohairsip:1000@176.249.0.77;tag=7f222672 To: sip:100@176.249.0.77;tag=as01491743 Call-ID: 2932f90ef302332b CSeq: 2 INVITE Server: Asterisk PBX 1.8.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:100@176.249.0.77:5060 Content-Type: application/sdp Content-Length: 258 v=0 o=root 1225456982 1225456982 IN IP4 176.249.0.77 s=Asterisk PBX 1.8.0 c=IN IP4 176.249.0.77 t=0 0 m=audio 15918 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv -- AGI Script Executing Application: (Playback) Options: (congestion,noanswer) -- SIP/1000-0019 Playing 'congestion.slin' (language 'en') -- SIP/1000-0019AGI Script agi.php completed,
[asterisk-users] TE410P (1st) without cables always green
Helo, I am upgrading a linux box ( Slackware + asterisk 1.0 + zaptel 0.9 ) to new asterisk 1.8 + dahdi. - In old softwares versions the box is working well. - After upgrade to slackware 13.37 + dahdi ( 2.3.X to SVN ) + asterisk , before asterisk is loaded and PRI/E1 Cables ( Span 1/2 ) + T1 ( Span 3/4 for Addtran channels banks ) , all Spans are in OK ( Green ) mode. - I go to another developer box, and the same problems cames up, is very dificult for me work in the productions box ( only 02:00 am to 04:00 am ). - Follwing my developer box setup : 1) System.conf ( I am working at this time with only 1 spam ) span=1,1,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 loadzone= us defaultzone = us 2) modprobe wct4xxp debug=3 root@zap4:~# dahdi_cfg -v DAHDI Tools Version - 2.3.0 DAHDI Version: 2.3.0.1 Echo Canceller(s): Configuration == SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet (DSX-1) 31 channels to configure. Setting echocan for channel 1 to mg2 Setting echocan for channel 2 to mg2 Setting echocan for channel 3 to mg2 Setting echocan for channel 4 to mg2 Setting echocan for channel 5 to mg2 Setting echocan for channel 6 to mg2 Setting echocan for channel 7 to mg2 Setting echocan for channel 8 to mg2 Setting echocan for channel 9 to mg2 Setting echocan for channel 10 to mg2 Setting echocan for channel 11 to mg2 Setting echocan for channel 12 to mg2 Setting echocan for channel 13 to mg2 Setting echocan for channel 14 to mg2 Setting echocan for channel 15 to mg2 Setting echocan for channel 16 to none Setting echocan for channel 17 to mg2 Setting echocan for channel 18 to mg2 Setting echocan for channel 19 to mg2 Setting echocan for channel 20 to mg2 Setting echocan for channel 21 to mg2 Setting echocan for channel 22 to mg2 Setting echocan for channel 23 to mg2 Setting echocan for channel 24 to mg2 Setting echocan for channel 25 to mg2 Setting echocan for channel 26 to mg2 Setting echocan for channel 27 to mg2 Setting echocan for channel 28 to mg2 Setting echocan for channel 29 to mg2 Setting echocan for channel 30 to mg2 Setting echocan for channel 31 to mg2 3) dahdi_scan dahdi_scan |more [1] active=yes alarms=OK = Here is the proble, the span is without cables. description=T4XXP (PCI) Card 0 Span 1 name=TE4/0/1 manufacturer=Digium devicetype=Wildcard TE410P/TE405P (1st Gen) location=Board ID Switch 0 basechan=1 totchans=31 irq=20 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS 4) dmesg , I put some informantion in bold. dahdi: Telephony Interface Unloaded dahdi: Telephony Interface Registered on major 196 dahdi: Version: 2.3.0.1 Found TE4XXP at base address e910, remapped to f895a000 DMA memory base of size 4096 at f6513000. Read: f6513800 and Write f6513000 TE4XXP version c01a009b, burst OFF card 0: FALC framer is v2.1 or earlier. FALC version: 0005, Board ID: 00 Reg 0: 0x36513800 Reg 1: 0x36513000 Reg 2: 0x07fc07fc Reg 3: 0x Reg 4: 0x Reg 5: 0x Reg 6: 0xc01a009b Reg 7: 0x1300 Reg 8: 0x010200ff Reg 9: 0x00fd Reg 10: 0x004a *IRQ 20/wct4xxp: IRQF_DISABLED is not guaranteed on shared IRQs wct4xxp :05:00.0: Enabled 1sec error counter interrupt wct4xxp :05:00.0: Enabled errored second interrupt wct4xxp :05:00.0: Enabled 1sec error counter interrupt wct4xxp :05:00.0: Enabled errored second interrupt wct4xxp :05:00.0: Enabled 1sec error counter interrupt wct4xxp :05:00.0: Enabled errored second interrupt wct4xxp :05:00.0: Enabled 1sec error counter interrupt wct4xxp :05:00.0: Enabled errored second interrupt *Found a Wildcard: Wildcard TE410P/TE405P (1st Gen) TE4XXP: Launching card: 0 TE4XXP: Setting up global serial parameters Successfully initialized serial bus for unit 0 Successfully initialized serial bus for unit 1 Successfully initialized serial bus for unit 2 Successfully initialized serial bus for unit 3 About to enter spanconfig! TE4XXP: Configuring span 1 Done with spanconfig! TE4XXP: Configured channel 1 (TE4/0/1/1) sigtype 128 dahdi_echocan_mg2: Registered echo canceler 'MG2' TE4XXP: Configured channel 2 (TE4/0/1/2) sigtype 128 TE4XXP: Configured channel 3 (TE4/0/1/3) sigtype 128 TE4XXP: Configured channel 4 (TE4/0/1/4) sigtype 128 TE4XXP: Configured channel 5 (TE4/0/1/5) sigtype 128 TE4XXP: Configured channel 6 (TE4/0/1/6) sigtype 128 TE4XXP: Configured channel 7 (TE4/0/1/7) sigtype 128 TE4XXP: Configured channel 8 (TE4/0/1/8) sigtype 128 TE4XXP: Configured channel 9 (TE4/0/1/9) sigtype 128 TE4XXP: Configured channel 10 (TE4/0/1/10) sigtype 128 TE4XXP: Configured channel 11 (TE4/0/1/11) sigtype 128 TE4XXP: Configured channel 12 (TE4/0/1/12) sigtype 128 TE4XXP: Configured channel 13 (TE4/0/1/13) sigtype 128 TE4XXP: Configured channel 14 (TE4/0/1/14) sigtype 128 TE4XXP: Configured channel 15 (TE4/0/1/15) sigtype 128
Re: [asterisk-users] Headset Options
Hi, Jabra headsets work fine with Polycom. Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af Blake Burgess Sendt: 7. februar 2012 05:01 Til: asterisk-users@lists.digium.com Emne: [asterisk-users] Headset Options Hey, I've heard recently from quite a few customers that there's cordless handsets around which don't require a lifter. Is anyone aware of any of these which will work with the cisco 69xx's, 79xx's or any of the current polycom range? -Blake -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] How to Send SMS on SS7 DChannel-16(Signaling Channel)
Dear All, i have created one SS7 link on E1 bye using DADHi/libss7/Asterisk1.8/DigiumTE205. Link is up and voice is working, but regarding SMS, Provider ask to send SMS on DChannel-16(Signaling Channel) by use below mentioned command but its giving error. ./smsq --motx-channel='DAHDI/16/+97333818181' 33297055 'Hello' -- Attempting call on DAHDI/16/+97333818181 for application SMS(0) (Retry 7) [Feb 8 11:20:54] NOTICE[18061]: channel.c:5322 __ast_request_and_dial: Unable to request channel DAHDI/16/+97333818181 [Feb 8 11:20:54] NOTICE[18061]: pbx_spool.c:353 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) kindly help me is it possible to send SMS on Signaling Channel and how? thanks. NOCTEST*CLIss7 show linkset 1 SS7 linkset 1 status: Up NOCTEST*CLI ss7 show channels link Chan Lcl Rem Call SS7 Channel set Chan Idle Blk Blk Level Call Name 11 Yes No No Idle No 12 Yes No No Idle No 13 Yes No No Idle No 14 Yes No No Idle No 15 Yes No No Idle No 16 Yes No No Idle No 17 Yes No No Idle No 18 Yes No No Idle No 19 Yes No No Idle No 1 10 Yes No No Idle No 1 11 Yes No No Idle No 1 12 Yes No No Idle No 1 13 Yes No No Idle No 1 14 Yes No No Idle No 1 15 Yes No No Idle No 1 17 Yes No No Idle No 1 18 Yes No No Idle No 1 19 Yes No No Idle No 1 20 Yes No No Idle No 1 21 Yes No No Idle No 1 22 Yes No No Idle No 1 23 Yes No No Idle No 1 24 Yes No No Idle No 1 25 Yes No No Idle No 1 26 Yes No No Idle No 1 27 Yes No No Idle No 1 28 Yes No No Idle No 1 29 Yes No No Idle No 1 30 Yes No No Idle No 1 31 Yes No No Idle No NOCTEST*CLI -- Regards, M. Asif Raza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Early Media configuration doesn't seem to be working
Hi We are using asterisk 1.8.7.0 Our Sip provider is passing us ringing via Early Media, i.e. using a SIP 183 Session Progress, with session description message which is fine for the most part but some of our customers are terminating on an ISDN gateway which doesn't interpret this message and those customers get no ringing. After doing some reading on the subject I have tried the following set prematuremedia=yes in sip.conf set progressinband=never in sip.conf set progressinband=never in the peers configuration in question but the asterisk server still passes on the 183 message and RTP stream rather than converting it to a SIP 180 Ringing message. Is there a problem here or am I misunderstanding something? Thanks in Advance Ish -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk V/s FreeSwitch
Hi List, Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk V/s FreeSwitch
You mean concurrent calls? You can have several 100 concurrent calls with a good CPU in newer versions of asterisk, however calls per secons (CPS) have some limitations I guess reason being that both are different in Architecture, Asterisk was designed keeping PBX in mind but Freeswitch was for SIP switching Regards, Zohair Raza On Tue, Feb 7, 2012 at 3:38 PM, virendra bhati virbh...@gmail.com wrote: Hi List, Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk V/s FreeSwitch
yes concurrent calls(CC). On Tue, Feb 7, 2012 at 5:27 PM, Zohair Raza engineerzuhairr...@gmail.comwrote: You mean concurrent calls? You can have several 100 concurrent calls with a good CPU in newer versions of asterisk, however calls per secons (CPS) have some limitations I guess reason being that both are different in Architecture, Asterisk was designed keeping PBX in mind but Freeswitch was for SIP switching Regards, Zohair Raza On Tue, Feb 7, 2012 at 3:38 PM, virendra bhati virbh...@gmail.com wrote: Hi List, Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy
Hello, I've been managing multiple call centres, almost all of them having their calls recorded one way or other. Even in PBX environments with MixMonitor and call recordings I haven't came across the situation where I discovered that I can't chanspy a call because its recorded ! Which version of asterisk you are using ! can you paste the CLI logs which show a complete call with a failed attempt to Chanspy ? Regards, Sammy On Tue, Feb 7, 2012 at 2:12 PM, Jonas Kellens jonas.kell...@telenet.bewrote: ** On 02/02/2012 11:24 AM, Jonas Kellens wrote: Hello, ChanSpy can not be used on a Channel that is being recorded with MixMonitor. How can I verify if a channel which I want to spy on, is currently not being recorded ?! Anyone with some feedback ?! I notice that ongoing recordings are temporarily saved in the directory /tmp. How could I look from the dialplan into the /tmp-directory to see if there is an ongoing recording for the channel that one wants to spy on ? Jonas. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Subscribe Problem - Zombie Channel
Hi Brian, Did you ever figure out what's causing this, and how to deal with it? I'm seeing the same behavior with call-pickups (it's rare, but it's happened a few times) on Asterisk 1.6.1.11 Did you figure out a way to get rid of the channel without restarting? Regards, Örn On Wed, Jul 28, 2010 at 9:45 PM, dotnetdub dotnet...@gmail.com wrote: On 28 July 2010 21:42, Stefan Schmidt s...@sil.at wrote: dotnetdub schrieb: Hi List, snip core show channels Channel Location State Application(Data) SIP/102--08e1 *8@from-inside Down (None) SIP/102--08d6 *8@from-inside Ring (None) SIP/102--08d7 *8@from-inside Ring (None) 3 active channels 0 active calls The only way to free them up is to force a restart. restart now Any clues on how I can debug this and try to sort it or even if anyone has come across this. Many thanks in advance. Brian hello, you should recompile asterisk with DEBUG CHANNEL LOCKS flag and i think you will see some locks when this happens with core show locks. how do you make the pickup? do you use an extension *8 for this, or just the feature for pickup in features conf? best regards steve Hi Steve, Thanks for the reply. We have: pickupexten = *8 ; Configure the pickup extension. Default is *8 in features.conf. I will recompile on one of the sites this happens on. It's really odd, can go for weeks without this happening and then a customer will report to me that their extension is showing in use and I will login and there can be two or three of these locks. On one site it actually makes asterisk impossible to stop and I need to kill -9 We have stuck with version 1.4.22 as it has been so solid for us, no dumps or deadlocks etc. We have tried to move to 1.4.25 and 1.4.29 but would experience random weirdness that we just don't get with this version. When recompiled with this flag and if indeed it does show locks, what would be the next step? Thanks for your help. Brian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy
On 02/07/2012 01:07 PM, Sammy Govind wrote: Hello, I've been managing multiple call centres, almost all of them having their calls recorded one way or other. Even in PBX environments with MixMonitor and call recordings I haven't came across the situation where I discovered that I can't chanspy a call because its recorded ! Which version of asterisk you are using ! can you paste the CLI logs which show a complete call with a failed attempt to Chanspy ? Using Asterisk 1.6.2.22. The fact that ChanSpy can not be used with MixMonitor is something I read on the wiki : Attention * Up to and including Asterisk 1.4.17 ChanSpy can cause a *crash/segfault* if used together with Monitor http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor or MixMonitor http://www.voip-info.org/wiki/view/MixMonitor at the same time. 1.4.18 is supposed to attack this issue by using audiohooks that replaces the current ChanSpy approach. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy
Oh Come on you are Using Asterisk 1.6.2.22. already. Atleast give it a shot and if this still persists then look for other methods or fixes. On Tue, Feb 7, 2012 at 5:44 PM, Jonas Kellens jonas.kell...@telenet.bewrote: ** On 02/07/2012 01:07 PM, Sammy Govind wrote: Hello, I've been managing multiple call centres, almost all of them having their calls recorded one way or other. Even in PBX environments with MixMonitor and call recordings I haven't came across the situation where I discovered that I can't chanspy a call because its recorded ! Which version of asterisk you are using ! can you paste the CLI logs which show a complete call with a failed attempt to Chanspy ? Using Asterisk 1.6.2.22. The fact that ChanSpy can not be used with MixMonitor is something I read on the wiki : Attention - Up to and including Asterisk 1.4.17 ChanSpy can cause a * crash/segfault* if used together with Monitorhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Monitoror MixMonitor http://www.voip-info.org/wiki/view/MixMonitor at the same time. 1.4.18 is supposed to attack this issue by using audiohooks that replaces the current ChanSpy approach. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy
that means that from 1.4.18 that issue is no longer present ? On 7 February 2012 12:44, Jonas Kellens jonas.kell...@telenet.be wrote: ** On 02/07/2012 01:07 PM, Sammy Govind wrote: Hello, I've been managing multiple call centres, almost all of them having their calls recorded one way or other. Even in PBX environments with MixMonitor and call recordings I haven't came across the situation where I discovered that I can't chanspy a call because its recorded ! Which version of asterisk you are using ! can you paste the CLI logs which show a complete call with a failed attempt to Chanspy ? Using Asterisk 1.6.2.22. The fact that ChanSpy can not be used with MixMonitor is something I read on the wiki : Attention - Up to and including Asterisk 1.4.17 ChanSpy can cause a * crash/segfault* if used together with Monitorhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Monitoror MixMonitor http://www.voip-info.org/wiki/view/MixMonitor at the same time. 1.4.18 is supposed to attack this issue by using audiohooks that replaces the current ChanSpy approach. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Custom extension: dial a queue
For dial a local extension in a queue Local/4555@extension,1,s Saludos Neri -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?
On Mon, 06 Feb 2012 10:24:42 -0600 (CST), Richard Mudgett rmudg...@digium.com wrote: The UPGRADE.txt and CHANGES files do just that. They have been a part of the Asterisk source files for a long time. Thanks for the info. The problem is that the ChangeLog files http://downloads.asterisk.org/pub/telephony/asterisk/releases/ are very long to read, and make no distinction between tiny features/bug fixes and major changes, so non-experts are unable to tell them apart. No Asterisk expert keeps track of new releases and blogs about major changes when they occur? At the very least, what is the main difference between the four branches currently under development, so that 1.4 users can tell if it's worth upgrading to another branch (save for the end-of-lifed branches)? Thank you. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?
On 7 Feb 2012, at 14:27, Gilles wrote: On Mon, 06 Feb 2012 10:24:42 -0600 (CST), Richard Mudgett rmudg...@digium.com wrote: The UPGRADE.txt and CHANGES files do just that. They have been a part of the Asterisk source files for a long time. Thanks for the info. The problem is that the ChangeLog files http://downloads.asterisk.org/pub/telephony/asterisk/releases/ are very long to read, and make no distinction between tiny features/bug fixes and major changes, so non-experts are unable to tell them apart. The upgrade files may be more to your tastes than changes files. There is no comparison chart that I know of. Just use the latest version that has a support-lifetime suitable to your needs. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?
On Tue, 7 Feb 2012 14:31:31 +, Steven Howes steve-li...@geekinter.net wrote: The upgrade files may be more to your tastes than changes files. Thanks. I downloaded and untarred asterisk-1.8.8.0.tar.gz, and it looks like the UPGRADE*.txt files within tarballs are the closest there is to knowing what major features were introduced in each branch, so as to make an educated guess as to whether it's worth upgrading to a newer release/branch. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Callmanager 4 Asterisk Malformed/Missing URL
Hi Becca, Thanks for that but that's actually the guide I have been following to get me this far! There seems to be a fair number of posts on the net from people with the same problem I just can't seem to find an answer ! It is now officially driving me nuts !! Nigel From: Rebecca Robinson rebecca.robin...@amgsrv.com To: Nigel Eldred nigel_eld...@yahoo.com; asterisk-users@lists.digium.com Sent: Monday, February 6, 2012 1:48 PM Subject: RE: [asterisk-users] Callmanager 4 Asterisk Malformed/Missing URL Nigel, I have never personally setup the Call Manager (CUCM), or whatever they call it today, to work with Asterisk. But I have seen what appears to be a good guide on voip-info.org. http://www.voip-info.org/wiki/view/Asterisk+Cisco+CallManager+Integration Becca From:asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nigel Eldred Sent: Monday, February 06, 2012 3:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Callmanager 4 Asterisk Malformed/Missing URL Hi, I am currently trying to get a Cisco Callmanager 4.1 and an Asterisk server (1.6.2.21) to talk via a SIP trunk so I can use the Voicemail component of the Asterisk (all the phones are associated with the Callmanager). The connection seem to be there. When I do a sip show peers on the Asterisk server I see the Callmanager as Monitored and online however I can't get any calls to pass from the CM to the Asterisk. If I debug the SIP I get a regular SIP/2.0 400 Bad Request - 'Malformed/Missing URL' which is from the Callmanager. Can anybody tell me the cause of this message and/or how I can resolve the problem ? Any help would be greatly appreciated Many thanks NigelConfidentiality Statement Notice: This email is covered by the Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and intended only for the use of the individual or entity to whom it is addressed. Any review, retransmission, dissemination to unauthorized persons or other use of the original message and any attachments is strictly prohibited. If you received this electronic transmission in error, please reply to the above-referenced sender about the error and permanently delete this message. Thank you for your cooperation.-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE410P (1st) without cables always green
On Tue, Feb 07, 2012 at 08:23:53AM -0200, Marcio Gomes wrote: - After upgrade to slackware 13.37 + dahdi ( 2.3.X to SVN ) + asterisk , before asterisk is loaded and PRI/E1 Cables ( Span 1/2 ) + T1 ( Span 3/4 for Addtran channels banks ) , all Spans are in OK ( Green ) mode. - I go to another developer box, and the same problems cames up, is very dificult for me work in the productions box ( only 02:00 am to 04:00 am ). - Follwing my developer box setup : snip 5) cat /proc/interrupts cat /proc/interrupts 1 CPU0 CPU1 0: 598466 686790 IO-APIC-edge timer 1: 4 4 IO-APIC-edge i8042 8: 0 0 IO-APIC-edge rtc0 9: 0 0 IO-APIC-fasteoi acpi 16: 0 0 IO-APIC-fasteoi pata_jmicron 19: 488966 400468 IO-APIC-fasteoi ata_piix, ata_piix 20: 20 8 IO-APIC-fasteoi wct4xxp 28: 3267 3254 PCI-MSI-edge eth0 NMI:12852981285102 Non-maskable interrupts LOC: 686792 598229 Local timer interrupts SPU: 0 0 Spurious interrupts PMI: 0 0 Performance monitoring interrupts PND: 0 0 Performance pending work RES:535493 Rescheduling interrupts CAL: 25 36 Function call interrupts TLB:136124 TLB shootdowns TRM: 0 0 Thermal event interrupts THR: 0 0 Threshold APIC interrupts MCE: 0 0 Machine check exceptions MCP: 5 5 Machine check polls ERR: 1 MIS: 0 It looks like your development box is having problems with interrupts from the card. Once you run dahdi_cfg for the span you should be getting 1 interrupts/sec and above I can see you only got 28. I've seen this recently with some risers which was the reason for commit 10380 wct4xxp: Fail startup if not generating interrupts. [1] [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10380 Do your development boxes work with an older version of DAHDI? Just not 2.3.0.1? Also, why not upgrade to 2.5.0.2 or the trunk of the 2.6 branch? Cheers, Shaun -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk V/s FreeSwitch
On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati virbh...@gmail.com wrote: Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... Provided Asterisk, even in release 1.8 or 10, does handle much fewer concurrent calls than Freeswitch, you might find the answer in those articles: How does FreeSWITCH compare to Asterisk? www.freeswitch.org/node/117 Asterisk vs FreeSWITCH www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ Asterisk vs. FreeSWITCH www.anders.com/cms/266 Open Source VoIP: Asterisk or FreeSwitch? www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233 FreeSwitch vs Asterisk www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk V/s FreeSwitch
Freeswitch was engineered from scratch by some Asterisk developers who wanted to start afresh on a cleaner programming base. Asterisk is like Topsy, She just growed and had to maintain backward compatibility. The latest versions of Asterisk are reported to be much improved in that respect. On 7 February 2012 15:40, Gilles codecompl...@free.fr wrote: On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati virbh...@gmail.com wrote: Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... Provided Asterisk, even in release 1.8 or 10, does handle much fewer concurrent calls than Freeswitch, you might find the answer in those articles: How does FreeSWITCH compare to Asterisk? www.freeswitch.org/node/117 Asterisk vs FreeSWITCH www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ Asterisk vs. FreeSWITCH www.anders.com/cms/266 Open Source VoIP: Asterisk or FreeSwitch? www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233 FreeSwitch vs Asterisk www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Le 06/02/2012 04:04, Gilles a écrit : Hello Is there a document that sums up the major changes made to the four main releases available (1.4, 1.6, 1.8, and 10), to check if it's worth upgrading? This link also presents changes between Asterisk versions: http://linuxinnovations.com/applications1.4-1.6.2.html Thanks, - -- Jean-Denis Girard SysNux Systèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 50 10 40 / GSM: +689 79 75 27 -BEGIN PGP SIGNATURE- iEYEARECAAYFAk8xTP0ACgkQuu7Rv+oOo/gmlgCeKdH/TPWOAM5cIG+Ee0L9cG1e exEAn3ThS2K+JUvxUNhcuNd3GAsdzZKq =6T+v -END PGP SIGNATURE- -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to Send SMS on SS7 DChannel-16(Signaling Channel)
Dear All, i have created one SS7 link on E1 bye using DADHi/libss7/Asterisk1.8/DigiumTE205. Link is up and voice is working, but regarding SMS, Provider ask to send SMS on DChannel-16(Signaling Channel) by use below mentioned command but its giving error. ./smsq --motx-channel='DAHDI/16/+97333818181' 33297055 'Hello' -- Attempting call on DAHDI/16/+97333818181 for application SMS(0) (Retry 7) [Feb 8 11:20:54] NOTICE[18061]: channel.c:5322 __ast_request_and_dial: Unable to request channel DAHDI/16/+97333818181 [Feb 8 11:20:54] NOTICE[18061]: pbx_spool.c:353 attempt_thread: Call failed to go through, reason (0) Call Failure (not BUSY, and not NO_ANSWER, maybe Circuit busy or down?) kindly help me is it possible to send SMS on Signaling Channel and how? thanks. I do not think that Asterisk supports SMS with SS7. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE410P (1st) without cables always green
On 07-02-12 16:30, Shaun Ruffell wrote: [snip] It looks like your development box is having problems with interrupts from the card. Once you run dahdi_cfg for the span you should be getting 1 interrupts/sec and above I can see you only got 28. I've seen this recently with some risers which was the reason for commit 10380 wct4xxp: Fail startup if not generating interrupts. [1] [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10380 To be honest the DAHDI startup failed: Input/output error error message provided by dahdi_cfg is not very helpful. Would it perhaps be an idea to use the message in the kernel log wct4xxp :02:08.0: Interrupts not detected. also in the error message from dahdi_cfg (at least the interrupts not detected part)? That would give a clue what's going on without having to dig through logfiles. Just my 0.02. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy
It's a good thing I never read that warning, since I've been using those in a call center environment for about seven years and never had that issue. Started with 1.2, went to 1.4 and 1.6 now. So I can't answer your question about when it was fixed but I've never had a problem doing it (70 concurrent calls max, all recorded, 5 concurrent channels spied max). On Tue, Feb 7, 2012 at 5:48 AM, Tiago Geada tiago.ge...@gmail.com wrote: that means that from 1.4.18 that issue is no longer present ? On 7 February 2012 12:44, Jonas Kellens jonas.kell...@telenet.be wrote: ** On 02/07/2012 01:07 PM, Sammy Govind wrote: Hello, I've been managing multiple call centres, almost all of them having their calls recorded one way or other. Even in PBX environments with MixMonitor and call recordings I haven't came across the situation where I discovered that I can't chanspy a call because its recorded ! Which version of asterisk you are using ! can you paste the CLI logs which show a complete call with a failed attempt to Chanspy ? Using Asterisk 1.6.2.22. The fact that ChanSpy can not be used with MixMonitor is something I read on the wiki : Attention - Up to and including Asterisk 1.4.17 ChanSpy can cause a * crash/segfault* if used together with Monitorhttp://www.voip-info.org/wiki/view/Asterisk+cmd+Monitoror MixMonitor http://www.voip-info.org/wiki/view/MixMonitor at the same time. 1.4.18 is supposed to attack this issue by using audiohooks that replaces the current ChanSpy approach. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy
Only trust the wiki if it explicitly refers to your current version (and then you should still test it). From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Carlos Alvarez Sent: Tuesday, February 07, 2012 10:46 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MixMonitor and ChanSpy It's a good thing I never read that warning, since I've been using those in a call center environment for about seven years and never had that issue. Started with 1.2, went to 1.4 and 1.6 now. So I can't answer your question about when it was fixed but I've never had a problem doing it (70 concurrent calls max, all recorded, 5 concurrent channels spied max). On Tue, Feb 7, 2012 at 5:48 AM, Tiago Geada tiago.ge...@gmail.com wrote: that means that from 1.4.18 that issue is no longer present ? On 7 February 2012 12:44, Jonas Kellens jonas.kell...@telenet.be wrote: On 02/07/2012 01:07 PM, Sammy Govind wrote: Hello, I've been managing multiple call centres, almost all of them having their calls recorded one way or other. Even in PBX environments with MixMonitor and call recordings I haven't came across the situation where I discovered that I can't chanspy a call because its recorded ! Which version of asterisk you are using ! can you paste the CLI logs which show a complete call with a failed attempt to Chanspy ? Using Asterisk 1.6.2.22. The fact that ChanSpy can not be used with MixMonitor is something I read on the wiki : Attention * Up to and including Asterisk 1.4.17 ChanSpy can cause a crash/segfault if used together with Monitor http://www.voip-info.org/wiki/view/Asterisk+cmd+Monitor or MixMonitor http://www.voip-info.org/wiki/view/MixMonitor at the same time. 1.4.18 is supposed to attack this issue by using audiohooks that replaces the current ChanSpy approach. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE410P (1st) without cables always green
On Tue, Feb 07, 2012 at 05:33:03PM +0100, Patrick Lists wrote: On 07-02-12 16:30, Shaun Ruffell wrote: [snip] It looks like your development box is having problems with interrupts from the card. Once you run dahdi_cfg for the span you should be getting 1 interrupts/sec and above I can see you only got 28. I've seen this recently with some risers which was the reason for commit 10380 wct4xxp: Fail startup if not generating interrupts. [1] [1] http://svnview.digium.com/svn/dahdi?view=revisionrevision=10380 To be honest the DAHDI startup failed: Input/output error error message provided by dahdi_cfg is not very helpful. Would it perhaps be an idea to use the message in the kernel log wct4xxp :02:08.0: Interrupts not detected. also in the error message from dahdi_cfg (at least the interrupts not detected part)? That would give a clue what's going on without having to dig through logfiles. Just my 0.02. Regards, Patrick I hear what you're saying although I don't think it's practical in this case. For drivers, when there is a general hardware failure the recommended practice is to return -EIO and log a message in the kernel log. The EIO is the clue that the log should be checked. Otherwise what typically happens is driver writers look through all the different error codes and try to find a one with a description that matches up with what they feel the problem is. The problem with this approach is that different error codes mean different things to different user mode code (EFAULT, ETOBIG, ENOSYS, EINTR, etc..). So one alternative would be for dahdi driver to have their own message log and then users of DAHDI interfaces could call an IOCTL to read the buffer on failure...but this is more code than necessary, IMO, for what should be a rare failure condition and these messages would also become set in stone as part of the API and couldn't change as necessary within a particular stable branch. The other alternative would be for dahdi_cfg to assume an EIO from a startup always means interrupts were failing, but this isn't future proof in case there are other hardware / platform failures that can be detected in the startup code. -- Shaun Ruffell Digium, Inc. | Linux Kernel Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] MixMonitor and ChanSpy
On Tue, Feb 7, 2012 at 9:47 AM, Danny Nicholas da...@debsinc.com wrote: Only trust the wiki if it explicitly refers to your current version (and then you should still test it). THIS. Believe him, when it comes to Asterisk, don't trust the docs, try it. Or read the code. There is no other way to really know. -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Headset Options
Jabra makes a wireless headset with a lifter-less remote answer?? Got a part number? We use the Plantronics CS-70 with the electronic lifter kit on Polycom phones. I don't know if there is anything for 79xx phones, we wouldn't use them if they were free. There is not one for Cisco SPA phones. On Tue, Feb 7, 2012 at 3:37 AM, Brynjolfur Thorvardsson bi...@itanet.nuwrote: Hi, Jabra headsets work fine with Polycom. ** ** *Fra:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *På vegne af *Blake Burgess *Sendt:* 7. februar 2012 05:01 *Til:* asterisk-users@lists.digium.com *Emne:* [asterisk-users] Headset Options ** ** Hey, ** ** I’ve heard recently from quite a few customers that there’s cordless handsets around which don’t require a lifter. ** ** Is anyone aware of any of these which will work with the cisco 69xx’s, 79xx’s or any of the current polycom range? ** ** -Blake -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding to 0.0.0.0 a security risk?
Steve Edwards, 2012-02-06 01:43: Unfortunately, (IIRC) Asterisk does not reply to the same interface packets are received from which limits the usefulness of multiple interfaces. Right, that's what I also observed. We had to take special measures to handle this. The problem lies in the nature of connectionless protocols as UDP. We also use freeradius, which does it right by itself (but still needs a compile time switch --with-udpfromto for it). -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding to 0.0.0.0 a security risk?
As far as I know, Asterisk would use the default Linux/Unix routing algorithms to send packets out, in which case yes: responses may not go out on the same interface packets were received on. E.g. if you receive packets with non-LAN IP addresses on eth0, while your default route is set to eth1, in the absence of custom routing Linux will send the responses over eth1. Thanks, another mystery solved then - Asterisk does rely on the Linux/Unix routing, in which case I would definitely need to take care of the SNAT/DNAT and proper routing/forwarding of packets between interfaces using core Linux/Unix tools. Am I correct in thinking that? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] res_http_post.so questions
The primary goal was to upload audio for IVRs in the Asterisk GUI. Thanks, if I don't use the GUI is it safe to exclude it from the build (it is just that I want to avoid a bunch of other dependencies which come with that module)? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is this doable?
It is indeed. This is already implemented in Asterisk I take it then? If so, brilliant news! More or less. I don't know if it's easy to trigger for specific caller ID values, or for none. You might need to to a little customization, but something mostly like what you describe is present. I am glad to see this! Which modules/functions present this functionality - do you know? I am almost certainly going to customise this as the screening of calls will be done using my own custom-defined criteria and the response options will also have to be customised/enhanced as well (how much really depends on what is currently implemented in Asterisk). Is there some kind of attack that you believe is possible on one interface that isn't on the other? I can't conceive of any way that making your service available on additional addresses increases your vulnerability. Of course it does - by making Asterisk service available on, say eth2 (by binding on 0.0.0.0 that is automatically enabled, i.e. Asterisk can receive packets coming from that interface). This is not what I want. If I could restrict Asterisk to bind only on the eth0 and eth1 for example, packets coming from that interface (eth2) won't affect Asterisk at all and they will either be dropped or rejected as nothing would listen on that address/port. I know that you may say netfilter/iptables is there to protect you, but the system will be more secure if Asterisk don't have the (physical) ability to answer requests coming from undesired interfaces - regardless of whether I have a fully-functional netfilter/iptables in place (even if it is compromised), rather than having Asterisk potentially answering such requests (by binding to 0.0.0.0) even if netfilter/iptables are functioning. In other words, having physically restricted Asterisk from answering requests coming from undesired interfaces (short of directly forwarding/routing packets from/to that interface) is better than allowing it do so and relying solely on netfilter/iptables for protection. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding to 0.0.0.0 a security risk?
All of that is true, but none of it appears to be a security concern, specifically. For you, may be, but from where I am sitting, I don't want to rely solely on netfilter/iptables to protect me when I could physically restrict Asterisk from binding to that interface (and answering such requests) - that will serve me well in the event netfilter/iptables is somehow compromised (see my previous post). It's possible for an application to bind a socket to a specific interface, but very few do. Generally speaking, server applications bind a socket to an address. The kernel decides what interface that packets are sent on. Normally that will be the interface that has the lowest cost default route, not necessarily the one on which a connection was initiated. That is why I noted previously that you have to use connection tracking, packet mangling, and ip rules for multi-homed hosts. If you've never verified that your packets are being routed out the interface you expect (probably with tcpdump), perhaps you should. Yeah, that was already clarified by another poster - I assumed (wrongly, as it turned out) that Asterisk, somehow, could automagically take care of directing sip/voip packets between interfaces and also take care of all the other related issues. As I understand it now, I will have to reconfigure this myself by using the standard Linux/Unix tools (ip iptables mostly). Thanks for the clarification yet again! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE410P (1st) without cables always green
On 07-02-12 17:54, Shaun Ruffell wrote: [snip] I hear what you're saying although I don't think it's practical in this case. [snip] Thank you for your elaborate feedback. It's clear why you are the one developing and I send my 0.02 to the mailinglist :) I see your point. Digging in logfiles it is. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding to 0.0.0.0 a security risk?
On 07-02-12 18:41, Josh wrote: [snip] Thanks, another mystery solved then - Asterisk does rely on the Linux/Unix routing, in which case I would definitely need to take care of the SNAT/DNAT and proper routing/forwarding of packets between interfaces using core Linux/Unix tools. Am I correct in thinking that? Yes. Regards, Patrick -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Headset Options
I have used a Plantronics CS351N and a CS70N with Polycom IP550 desk units. (both are single-ear units, in different forms) Each one needed a Plantronics APP-5 to replace using a lifter. They worked fine. The one complaint that I had from users is that the headset beep to show that a call was ringing in was not adjustable in volume, so it could be startling. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Blake Burgess Sent: Monday, February 06, 2012 11:01 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] Headset Options Hey, I've heard recently from quite a few customers that there's cordless handsets around which don't require a lifter. Is anyone aware of any of these which will work with the cisco 69xx's, 79xx's or any of the current polycom range? -Blake -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE410P (1st) without cables always green
Hello Shaun, Thanks for your answer, I try all dahdi I can compile in Slackware 13.1 and 13.37 ( recently I downgrade to 13.1, to compile some older kernels, but it is not the answer. ) It looks like your development box is having problems with interrupts from the card. Once you run dahdi_cfg for the span you should be getting 1 interrupts/sec and above I can see you only got 28. Yeah.. I forget to look at it. As i say, I try with all.. let's go.. I recompile the dahdi-linux-complete-2.6.0+2.6.0 witj 2.6.33.4 kernel. The interrupt problems persists. 20:630 1476 IO-APIC-fasteoi wct4xxp Do your development boxes work with an older version of DAHDI? Just not 2.3.0.1? Also, why not upgrade to 2.5.0.2 or the trunk of the 2.6 branch? All versions 2.2.XXX to SVN same problems... This is interrupt outputs to today SVN : 20: 1348 1788 IO-APIC-fasteoi wct4xxp ** This warning message, can be ignored in compilation ? from /usr/src/ASTERISK/20111212/DAHDI/dahdi-kernel/drivers/dahdi/dahdi-base.c:67: In function 'copy_from_user', inlined from 'dahdi_chan_write' at /usr/src/ASTERISK/20111212/DAHDI/dahdi-kernel/drivers/dahdi/dahdi-base.c:2449: /usr/src/linux-2.6.33.4/arch/x86/include/asm/uaccess_32.h:212: warning: call to 'copy_from_user_overflow' declared with attribute warning: copy_from_user() buffer size is not provably correct CC [M] /usr/src/ASTERISK/20111212/DAHDI/dahdi-kernel/drivers/dahdi/dahdi-sysfs.o ** In production box a intersting cat /proc/interrupts, * this is the working setup # cat /usr/src/ASTERISK/COMPILA/zaptel/ChangeLog 0.1.6: * Move network structures to be malloc()'d when needed * Add HDLC PPP Support * Fix multi-channel stuff in zaptel and torisa uname -a Linux zap1 2.6.11.12-ul2 #6 SMP Mon Dec 14 17:40:08 BRST 2009 i686 unknown unknown GNU/Linux zap1*CLI show version Asterisk CVS-HEAD-11/14/05-18:16:29 built by root@zap1 on a i686 running Linux cat /proc/interrupts CPU0 CPU1 0: 15203536 0IO-APIC-edge timer 8: 2 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 14: 42471 0IO-APIC-edge ide0 15:175 0IO-APIC-edge ide1 169: 15161016 0 IO-APIC-level libata, wcfxo 177: 130432 0 IO-APIC-level eth0 193: 15159692 0 IO-APIC-level t4xxp 201: 15160988 0 IO-APIC-level wcfxo NMI: 0 0 LOC: 15203154 15203153 ERR: 0 MIS: 0 * this is the NEW setup , the system has 2 hardisks with 2 slacks, 2 asterisk setups.. all compiled from sources ( I not like pre build softwares , when i am in trouble I have serious problems to modify, and slackware has poor packages but is lovely stable distrib ) cat /proc/interrupts ( this kernel are without SMP compilation, but the problem is the same if I compile with SMP ) root@zap2:/etc/dahdi# cat /proc/interrupts CPU0 0: 73 IO-APIC-edge timer 1: 2 IO-APIC-edge i8042 2: 0XT-PIC-XTcascade 9: 4 IO-APIC-fasteoi 12: 4 IO-APIC-edge i8042 14: 1476 IO-APIC-edge ata_piix 15: 8583 IO-APIC-edge ata_piix 18: 62007 IO-APIC-fasteoi ata_piix, wcfxo 19: 62119 IO-APIC-fasteoi wcfxo 21: 1022 IO-APIC-fasteoi eth0 22: 1039 IO-APIC-fasteoi wct4xxp NMI: 0 Non-maskable interrupts LOC: 20739 Local timer interrupts SPU: 0 Spurious interrupts PMI: 0 Performance monitoring interrupts PND: 0 Performance pending work RES: 0 Rescheduling interrupts CAL: 0 Function call interrupts TLB: 0 TLB shootdowns TRM: 0 Thermal event interrupts THR: 0 Threshold APIC interrupts MCE: 0 Machine check exceptions MCP: 1 Machine check polls ERR: 0 In new and old SPAN 2 is without cable. [2] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 2 name=TE4/0/2 manufacturer=Digium devicetype=Wildcard TE410P/TE405P (1st Gen) location=Board ID Switch 0 basechan=32 totchans=31 irq=22 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS/CRC4 I really not understand de APIC changes in IO-APIC- and the greater 100 Interrupt numbers in old setups.. can you help me ? regards, marcio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE410P (1st) without cables always green
Hello Shaun, Thanks for your answer, I try all dahdi I can compile in Slackware 13.1 and 13.37 ( recently I downgrade to 13.1, to compile some older kernels, but it is not the answer. ) It looks like your development box is having problems with interrupts from the card. Once you run dahdi_cfg for the span you should be getting 1 interrupts/sec and above I can see you only got 28. Yeah.. I forget to look at it. As i say, I try with all.. let's go.. I recompile the dahdi-linux-complete-2.6.0+2.6.0 witj 2.6.33.4 kernel. The interrupt problems persists. 20:630 1476 IO-APIC-fasteoi wct4xxp Do your development boxes work with an older version of DAHDI? Just not 2.3.0.1? Also, why not upgrade to 2.5.0.2 or the trunk of the 2.6 branch? All versions 2.2.XXX to SVN same problems... This is interrupt outputs to today SVN : 20: 1348 1788 IO-APIC-fasteoi wct4xxp ** This warning message, can be ignored in compilation ? from /usr/src/ASTERISK/20111212/DAHDI/dahdi-kernel/drivers/dahdi/dahdi-base.c:67: In function 'copy_from_user', inlined from 'dahdi_chan_write' at /usr/src/ASTERISK/20111212/DAHDI/dahdi-kernel/drivers/dahdi/dahdi-base.c:2449: /usr/src/linux-2.6.33.4/arch/x86/include/asm/uaccess_32.h:212: warning: call to 'copy_from_user_overflow' declared with attribute warning: copy_from_user() buffer size is not provably correct CC [M] /usr/src/ASTERISK/20111212/DAHDI/dahdi-kernel/drivers/dahdi/dahdi-sysfs.o ** In production box a intersting cat /proc/interrupts, * this is the working setup # cat /usr/src/ASTERISK/COMPILA/zaptel/ChangeLog 0.1.6: * Move network structures to be malloc()'d when needed * Add HDLC PPP Support * Fix multi-channel stuff in zaptel and torisa uname -a Linux zap1 2.6.11.12-ul2 #6 SMP Mon Dec 14 17:40:08 BRST 2009 i686 unknown unknown GNU/Linux zap1*CLI show version Asterisk CVS-HEAD-11/14/05-18:16:29 built by root@zap1 on a i686 running Linux cat /proc/interrupts CPU0 CPU1 0: 15203536 0IO-APIC-edge timer 8: 2 0IO-APIC-edge rtc 9: 0 0 IO-APIC-level acpi 14: 42471 0IO-APIC-edge ide0 15:175 0IO-APIC-edge ide1 169: 15161016 0 IO-APIC-level libata, wcfxo 177: 130432 0 IO-APIC-level eth0 193: 15159692 0 IO-APIC-level t4xxp 201: 15160988 0 IO-APIC-level wcfxo NMI: 0 0 LOC: 15203154 15203153 ERR: 0 MIS: 0 * this is the NEW setup , the system has 2 hardisks with 2 slacks, 2 asterisk setups.. all compiled from sources ( I not like pre build softwares , when i am in trouble I have serious problems to modify, and slackware has poor packages but is lovely stable distrib ) cat /proc/interrupts ( this kernel are without SMP compilation, but the problem is the same if I compile with SMP ) root@zap2:/etc/dahdi# cat /proc/interrupts CPU0 0: 73 IO-APIC-edge timer 1: 2 IO-APIC-edge i8042 2: 0XT-PIC-XTcascade 9: 4 IO-APIC-fasteoi 12: 4 IO-APIC-edge i8042 14: 1476 IO-APIC-edge ata_piix 15: 8583 IO-APIC-edge ata_piix 18: 62007 IO-APIC-fasteoi ata_piix, wcfxo 19: 62119 IO-APIC-fasteoi wcfxo 21: 1022 IO-APIC-fasteoi eth0 22: 1039 IO-APIC-fasteoi wct4xxp NMI: 0 Non-maskable interrupts LOC: 20739 Local timer interrupts SPU: 0 Spurious interrupts PMI: 0 Performance monitoring interrupts PND: 0 Performance pending work RES: 0 Rescheduling interrupts CAL: 0 Function call interrupts TLB: 0 TLB shootdowns TRM: 0 Thermal event interrupts THR: 0 Threshold APIC interrupts MCE: 0 Machine check exceptions MCP: 1 Machine check polls ERR: 0 In new and old SPAN 2 is without cable. [2] active=yes alarms=OK description=T4XXP (PCI) Card 0 Span 2 name=TE4/0/2 manufacturer=Digium devicetype=Wildcard TE410P/TE405P (1st Gen) location=Board ID Switch 0 basechan=32 totchans=31 irq=22 type=digital-E1 syncsrc=0 lbo=0 db (CSU)/0-133 feet (DSX-1) coding_opts=HDB3 framing_opts=CCS,CRC4 coding=HDB3 framing=CCS/CRC4 I really not understand de APIC changes in IO-APIC- and the greater 100 Interrupt numbers in old setups.. can you help me ? regards, marcio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE410P (1st) without cables always green
Hello All, I really not understand de APIC changes in IO-APIC- and the greater 100 Interrupt numbers in old setups.. can you help me ? I forget to say, In production BOX, I only do a linux -R newconfig reboot , slots , hardware, setup, etc.. is the same.. no changes. It's give me a confirmation about it is not a hardware issue, the only changes are in software. In the development box, a hardware problem iss possible, I will make a copy of the working software and put in it to clarify this doubt. Regards, Marcio Gomes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8.9.1 Now Available
The Asterisk Development Team has announced the release of Asterisk 1.8.9.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.8.9.1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Fixes deadlocks occuring in chan_agent --- (Closes issue ASTERISK-19285. Reported by: Alex Villacis Lasso) * --- Ensure entering T.38 passthrough does not cause an infinite loop --- (Closes issue ASTERISK-18951. Reported-by: Kristijan Vrban) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.1 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 10.1.1 Now Available
The Asterisk Development Team has announced the release of Asterisk 10.1.1. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 10.1.1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following is a sample of the issues resolved in this release: * --- Fixes deadlocks occuring in chan_agent --- (Closes issue ASTERISK-19285. Reported by: Alex Villacis Lasso) * --- Ensure entering T.38 passthrough does not cause an infinite loop --- (Closes issue ASTERISK-18951. Reported-by: Kristijan Vrban) For a full list of changes in this release, please see the ChangeLog: http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.1 Thank you for your continued support of Asterisk! -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] TE410P (1st) without cables always green
News from Zaptel.. I patch the last zaptel.1.2.27 to compiles in 2.6.33 kernels series Surprise !! 20: 68011 71871 IO-APIC-fasteoi wct4xxp in zttool the spans are RED, all is working with pre 1.4 zaptel and dahdi. The problems are in changes between zaptel.1.2.X and zaptel.1.4.X. I thinking now that is an old issue added from 1.2 to 1.4 branches and sucessivily to dahdi branches, and I lost something in past. But I want go to asterisk 1.8 and greater with new drivers , Can anyone help me in this ? regards, Em 07/02/2012 17:05, Marcio Gomes escreveu: Hello All, I really not understand de APIC changes in IO-APIC- and the greater 100 Interrupt numbers in old setups.. can you help me ? I forget to say, In production BOX, I only do a linux -R newconfig reboot , slots , hardware, setup, etc.. is the same.. no changes. It's give me a confirmation about it is not a hardware issue, the only changes are in software. In the development box, a hardware problem iss possible, I will make a copy of the working software and put in it to clarify this doubt. Regards, Marcio Gomes -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Subscribe Problem - Zombie Channel
On Tuesday, 7 February 2012, Örn Arnarson o...@arnarson.net wrote: Hi Brian, Did you ever figure out what's causing this, and how to deal with it? I'm seeing the same behavior with call-pickups (it's rare, but it's happened a few times) on Asterisk 1.6.1.11 Did you figure out a way to get rid of the channel without restarting? Regards, Hi Orn I didn't find a way except a restart once active calls drop to zero. Regards, Brian On Wed, Jul 28, 2010 at 9:45 PM, dotnetdub dotnet...@gmail.com wrote: On 28 July 2010 21:42, Stefan Schmidt s...@sil.at wrote: dotnetdub schrieb: Hi List, snip core show channels Channel Location State Application(Data) SIP/102--08e1 *8@from-inside Down(None) SIP/102--08d6 *8@from-inside Ring(None) SIP/102--08d7 *8@from-inside Ring(None) 3 active channels 0 active calls The only way to free them up is to force a restart. restart now Any clues on how I can debug this and try to sort it or even if anyone has come across this. Many thanks in advance. Brian hello, you should recompile asterisk with DEBUG CHANNEL LOCKS flag and i think you will see some locks when this happens with core show locks. how do you make the pickup? do you use an extension *8 for this, or just the feature for pickup in features conf? best regards steve Hi Steve, Thanks for the reply. We have: pickupexten = *8; Configure the pickup extension. Default is *8 in features.conf. I will recompile on one of the sites this happens on. It's really odd, can go for weeks without this happening and then a customer will report to me that their extension is showing in use and I will login and there can be two or three of these locks. On one site it actually makes asterisk impossible to stop and I need to kill -9 We have stuck with version 1.4.22 as it has been so solid for us, no dumps or deadlocks etc. We have tried to move to 1.4.25 and 1.4.29 but would experience random weirdness that we just don't get with this version. When recompiled with this flag and if indeed it does show locks, what would be the next step? Thanks for your help. Brian -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Major changes between 1.4/1.6.1.8/10?
On Tue, 07 Feb 2012 06:10:37 -1000, Jean-Denis Girard jd.gir...@sysnux.pf wrote: This link also presents changes between Asterisk versions: http://linuxinnovations.com/applications1.4-1.6.2.html Thanks for the link. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding to 0.0.0.0 a security risk?
On 07/02/12 05:29, Gordon Messmer wrote: On 02/06/2012 03:27 PM, Josh wrote: Why do you see binding to 0.0.0.0 to be a security risk? Purely because a response from Asterisk can be received as a result of a connection on *any* interface on the system/machine. If I have Asterisk confined to, say, 2 interfaces - eth0 (10.1.1.1) and eth1 (10.2.1.1) then a request over a third/subsequent interface cannot be served - it is not normally possible. When Asterisk binds to 0.0.0.0 that is not the case and request over a third/subsequent interface *can* be served by Asterisk (provided the routing is setup properly, that is). All of that is true, but none of it appears to be a security concern, specifically. If you are connecting to the public internet, then it is much more important to think about a) do you really expose your Asterisk directly, or hide it behind a SIP router such as Kamailio? b) should you be using TLS (which is connection oriented and secured with certificates) rather than UDP? Everyone who connects with a cert has been screened in some way by a CA. c) if using TLS (or even just TCP), why not have the extra security of a port-forwarding from a firewall to the Asterisk TLS port? Then no other ports or addresses on the Asterisk box are exposed. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Binding to 0.0.0.0 a security risk?
On Tuesday 07 Feb 2012, Jakob Hirsch wrote: Steve Edwards, 2012-02-06 01:43: Unfortunately, (IIRC) Asterisk does not reply to the same interface packets are received from which limits the usefulness of multiple interfaces. Right, that's what I also observed. We had to take special measures to handle this. The problem lies in the nature of connectionless protocols as UDP. We also use freeradius, which does it right by itself (but still needs a compile time switch --with-udpfromto for it). Packets not going out on the same interface as the one they were received on is a general IP issue, not just for connectionless protocols. The same behaviour can be seen with TCP too. Unless you mangle with iptables or something, all information about the received interface has been stripped from the packet by the time it reaches the IP layer. /nitpick Regards, -- Raj -- Raj Mathur || r...@kandalaya.org || GPG: http://otheronepercent.blogspot.com || http://kandalaya.org || CC68 It is the mind that moves || http://schizoid.in || D17F -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk V/s FreeSwitch
thanks Gilles, After reading these web links. it's pretty clear that FreeSwitch is batter then Asterisk feature, quality wise. But asterisk is easy to used. But the question is still open from my end. *How* *FreeSwitch can support 1000CC but asterisk not* ? Because FreeSwitch used XML as configuration and asterisk plan text file ? FreeSwitch used sofia_sip and asterisk used sip ? Asterisk is PBX and FreeSwitch is SoftSwitch ? On Tue, Feb 7, 2012 at 9:10 PM, Gilles codecompl...@free.fr wrote: On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati virbh...@gmail.com wrote: Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... Provided Asterisk, even in release 1.8 or 10, does handle much fewer concurrent calls than Freeswitch, you might find the answer in those articles: How does FreeSWITCH compare to Asterisk? www.freeswitch.org/node/117 Asterisk vs FreeSWITCH www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ Asterisk vs. FreeSWITCH www.anders.com/cms/266 Open Source VoIP: Asterisk or FreeSwitch? www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233 FreeSwitch vs Asterisk www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk V/s FreeSwitch
Virendra- After reading these web links. it's pretty clear that FreeSwitch is batter then Asterisk feature, quality wise. But asterisk is easy to used. But the question is still open from my end. *How* *FreeSwitch can support 1000CC but asterisk not* ? Can you define your concurrent calls? IP-to-IP, or TDM-to-IP? I assume all G.711. Please specify. -Jeff Because FreeSwitch used XML as configuration and asterisk plan text file ? FreeSwitch used sofia_sip and asterisk used sip ? Asterisk is PBX and FreeSwitch is SoftSwitch ? On Tue, Feb 7, 2012 at 9:10 PM, Gilles codecompl...@free.fr wrote: On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati virbh...@gmail.com wrote: Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... Provided Asterisk, even in release 1.8 or 10, does handle much fewer concurrent calls than Freeswitch, you might find the answer in those articles: How does FreeSWITCH compare to Asterisk? www.freeswitch.org/node/117 Asterisk vs FreeSWITCH www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ Asterisk vs. FreeSWITCH www.anders.com/cms/266 Open Source VoIP: Asterisk or FreeSwitch? www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233 FreeSwitch vs Asterisk www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asterisk -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is this doable?
On 02/07/2012 09:43 AM, Josh wrote: More or less. I don't know if it's easy to trigger for specific caller ID values, or for none. You might need to to a little customization, but something mostly like what you describe is present. I am glad to see this! Which modules/functions present this functionality - do you know? http://www.asterisk.org/astdocs/node66.html Is there some kind of attack that you believe is possible on one interface that isn't on the other? I can't conceive of any way that making your service available on additional addresses increases your vulnerability. Of course it does - by making Asterisk service available on, say eth2 (by binding on 0.0.0.0 that is automatically enabled, i.e. Asterisk can receive packets coming from that interface). This is not what I want. Yes, I understand that it's not what you want, but that doesn't make it a security concern. If Asterisk is publicly available on one interface, making it available on another interface doesn't make you less secure. It's fine if you want to take that step, but please drop the everyone knows this is a security risk thing. You appear to be alone in that opinion, and unable to explain why you think it's a security risk. Moreover, you're speaking for others without warrant or welcome. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk V/s FreeSwitch
According to this article here: http://anders.com/cms/266 the difference mainly lies in how FreeSwitchs handles open channels in comparison with Asterisk. FS uses one thread per channel while * keeps jumping between threads. At least that's how I understand it. Fra: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] På vegne af virendra bhati Sendt: 8. februar 2012 06:34 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [asterisk-users] Asterisk V/s FreeSwitch thanks Gilles, After reading these web links. it's pretty clear that FreeSwitch is batter then Asterisk feature, quality wise. But asterisk is easy to used. But the question is still open from my end. How FreeSwitch can support 1000CC but asterisk not ? Because FreeSwitch used XML as configuration and asterisk plan text file ? FreeSwitch used sofia_sip and asterisk used sip ? Asterisk is PBX and FreeSwitch is SoftSwitch ? On Tue, Feb 7, 2012 at 9:10 PM, Gilles codecompl...@free.frmailto:codecompl...@free.fr wrote: On Tue, 7 Feb 2012 17:08:18 +0530, virendra bhati virbh...@gmail.commailto:virbh...@gmail.com wrote: Why FreeSwitch can handle more then 1,000CC and asterisk only 25CC ? What technology FreeSwitch is used and asterisk don't. I don't know it's the right or wrong but this question come to my mind... Provided Asterisk, even in release 1.8 or 10, does handle much fewer concurrent calls than Freeswitch, you might find the answer in those articles: How does FreeSWITCH compare to Asterisk? www.freeswitch.org/node/117http://www.freeswitch.org/node/117 Asterisk vs FreeSWITCH www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/http://www.richappsconsulting.com/blog/blog-detail/asterisk-vs-freeswitch/ Asterisk vs. FreeSWITCH www.anders.com/cms/266http://www.anders.com/cms/266 Open Source VoIP: Asterisk or FreeSwitch? www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233http://www.zdnet.com/blog/greenfield/open-source-voip-asterisk-or-freeswitch/233 FreeSwitch vs Asterisk www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asteriskhttp://www.dslreports.com/forum/r23246683-FreeSwitch-vs-Asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbh...@gmail.commailto:virbh...@gmail.com Skype id:- virbhati2 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
