[asterisk-users] DTMF recognized after call establishment
Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on SIP/MyTrunk-000a4b49, duration 0 ms [May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin '*' on SIP/MyTrunk-000a4b49 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on SIP/MyTrunk-000a4b49, duration 0 ms [May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin '8' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on SIP/MAN-000a4af0, duration 100 ms [May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8' with duration 100 queued on SIP/MAN-000a4af0 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8' queued on SIP/MAN-000a4af0 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on SIP/MAN-000a4b41, duration 100 ms [May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1' with duration 100 queued on SIP/MAN-000a4b41 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1' queued on SIP/MAN-000a4b41 [May 17 00:33:55] VERBOSE[4106] pbx.c: == Spawn extension (sip-trunk-inbound, 2127773456, 1) exited non-zero on 'SIP/MyTrunk-000a4af3' [May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing [h@trunk-outbound:1] NoOp(SIP/MAN-000a4b09, 16) in new stack [May 17 00:33:56] VERBOSE[4136] pbx.c: == Spawn extension (trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09' Is this some thing related to SIP RE-INVITE? Thanks. -- _ -- 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] Not able to build the chan_sip.c module
hi, anyone can help me to debug this ?? -- upendar On Mon, May 27, 2013 at 4:09 PM, upendra uppi...@gmail.com wrote: hi, chan_local and res_crypto are building but the chan_sip is not building . installed openssl also but still the chan_sip not building. -- Upendra On Mon, May 27, 2013 at 2:59 PM, Alec Davis siva...@paradise.net.nzwrote: i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see in make menuselect options it showing XXX -- extended , please let me know how to enable it and make build chan_sip module. -- Upendra from makeselect you'll find chan_sip depends on the following Depends on: chan_local(M), res_crypto(M), res_http_websocket(M) then you'll find res_crytpo is dependant on open_ssl Depends on: openssl(E) which for me on debian wheezy is libssl-dev Alec Davis -- _ -- 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] Not able to build the chan_sip.c module
Without posting exact error messages, dont expect help !! Mitul On May 28, 2013 1:01 PM, upendra uppi...@gmail.com wrote: hi, anyone can help me to debug this ?? -- upendar On Mon, May 27, 2013 at 4:09 PM, upendra uppi...@gmail.com wrote: hi, chan_local and res_crypto are building but the chan_sip is not building . installed openssl also but still the chan_sip not building. -- Upendra On Mon, May 27, 2013 at 2:59 PM, Alec Davis siva...@paradise.net.nzwrote: i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see in make menuselect options it showing XXX -- extended , please let me know how to enable it and make build chan_sip module. -- Upendra from makeselect you'll find chan_sip depends on the following Depends on: chan_local(M), res_crypto(M), res_http_websocket(M) then you'll find res_crytpo is dependant on open_ssl Depends on: openssl(E) which for me on debian wheezy is libssl-dev Alec Davis -- _ -- 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] Not able to build the chan_sip.c module
hi, there is no build errors , but the thing is that on Elastix Machine i want to install asterisk1.8.11.0 , while make the chan_sip module is not building, and when i see in the memuselect the chan_sip module driver showing as XXX to enable for building. -- Upendra. On Tue, May 28, 2013 at 1:03 PM, Mitul Limbani mi...@enterux.in wrote: Without posting exact error messages, dont expect help !! Mitul On May 28, 2013 1:01 PM, upendra uppi...@gmail.com wrote: hi, anyone can help me to debug this ?? -- upendar On Mon, May 27, 2013 at 4:09 PM, upendra uppi...@gmail.com wrote: hi, chan_local and res_crypto are building but the chan_sip is not building . installed openssl also but still the chan_sip not building. -- Upendra On Mon, May 27, 2013 at 2:59 PM, Alec Davis siva...@paradise.net.nzwrote: i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see in make menuselect options it showing XXX -- extended , please let me know how to enable it and make build chan_sip module. -- Upendra from makeselect you'll find chan_sip depends on the following Depends on: chan_local(M), res_crypto(M), res_http_websocket(M) then you'll find res_crytpo is dependant on open_ssl Depends on: openssl(E) which for me on debian wheezy is libssl-dev Alec Davis -- _ -- 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 -- _ -- 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] Not able to build the chan_sip.c module
Why not install the updated rpm version? Mitul On May 28, 2013 1:12 PM, upendra uppi...@gmail.com wrote: hi, there is no build errors , but the thing is that on Elastix Machine i want to install asterisk1.8.11.0 , while make the chan_sip module is not building, and when i see in the memuselect the chan_sip module driver showing as XXX to enable for building. -- Upendra. On Tue, May 28, 2013 at 1:03 PM, Mitul Limbani mi...@enterux.in wrote: Without posting exact error messages, dont expect help !! Mitul On May 28, 2013 1:01 PM, upendra uppi...@gmail.com wrote: hi, anyone can help me to debug this ?? -- upendar On Mon, May 27, 2013 at 4:09 PM, upendra uppi...@gmail.com wrote: hi, chan_local and res_crypto are building but the chan_sip is not building . installed openssl also but still the chan_sip not building. -- Upendra On Mon, May 27, 2013 at 2:59 PM, Alec Davis siva...@paradise.net.nzwrote: i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see in make menuselect options it showing XXX -- extended , please let me know how to enable it and make build chan_sip module. -- Upendra from makeselect you'll find chan_sip depends on the following Depends on: chan_local(M), res_crypto(M), res_http_websocket(M) then you'll find res_crytpo is dependant on open_ssl Depends on: openssl(E) which for me on debian wheezy is libssl-dev Alec Davis -- _ -- 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 -- _ -- 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] Not able to build the chan_sip.c module
On Mon, May 27, 2013 at 04:09:06PM +0530, upendra wrote: hi, chan_local and res_crypto are building but the chan_sip is not building . installed openssl also but still the chan_sip not building. ./menuselect/contrib/menuselect-dummy -c ./menuselect/contrib/menuselect-dummy -m sip -v What's the output? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Not able to build the chan_sip.c module
hi all, After installing packages openssl and openssl-devel packages the chan_sip is building . :) :) thanks to all for ur help. -- Upendra. On Tue, May 28, 2013 at 1:33 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Mon, May 27, 2013 at 04:09:06PM +0530, upendra wrote: hi, chan_local and res_crypto are building but the chan_sip is not building . installed openssl also but still the chan_sip not building. ./menuselect/contrib/menuselect-dummy -c ./menuselect/contrib/menuselect-dummy -m sip -v What's the output? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir -- _ -- 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] Not able to build the chan_sip.c module
please provide more information. how you are try to build asterisk, what is output of configure. witch headers configure script not found etc. On Tue, May 28, 2013 at 9:29 AM, upendra uppi...@gmail.com wrote: hi, anyone can help me to debug this ?? -- upendar On Mon, May 27, 2013 at 4:09 PM, upendra uppi...@gmail.com wrote: hi, chan_local and res_crypto are building but the chan_sip is not building . installed openssl also but still the chan_sip not building. -- Upendra On Mon, May 27, 2013 at 2:59 PM, Alec Davis siva...@paradise.net.nzwrote: i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see in make menuselect options it showing XXX -- extended , please let me know how to enable it and make build chan_sip module. -- Upendra from makeselect you'll find chan_sip depends on the following Depends on: chan_local(M), res_crypto(M), res_http_websocket(M) then you'll find res_crytpo is dependant on open_ssl Depends on: openssl(E) which for me on debian wheezy is libssl-dev Alec Davis -- _ -- 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] DTMF recognized after call establishment
i had this in past there was an ATA configured to send 9 at the end of dialing in my case. On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on SIP/MyTrunk-000a4b49, duration 0 ms [May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin '*' on SIP/MyTrunk-000a4b49 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on SIP/MyTrunk-000a4b49, duration 0 ms [May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin '8' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on SIP/MAN-000a4af0, duration 100 ms [May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8' with duration 100 queued on SIP/MAN-000a4af0 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8' queued on SIP/MAN-000a4af0 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on SIP/MAN-000a4b41, duration 100 ms [May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1' with duration 100 queued on SIP/MAN-000a4b41 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1' queued on SIP/MAN-000a4b41 [May 17 00:33:55] VERBOSE[4106] pbx.c: == Spawn extension (sip-trunk-inbound, 2127773456, 1) exited non-zero on 'SIP/MyTrunk-000a4af3' [May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing [h@trunk-outbound:1] NoOp(SIP/MAN-000a4b09, 16) in new stack [May 17 00:33:56] VERBOSE[4136] pbx.c: == Spawn extension (trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09' Is this some thing related to SIP RE-INVITE? Thanks. -- _ -- 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] G.729 codec in pass-thru mode
hello, 201.xxx.xxx.xxx = SIP Softphone which originates the call xxx.xxx.xxx.xxx = Asterisk server yyy.yyy.yyy.yyy = ITSP --- SIP read from UDP:201.xxx.xxx.xxx:5060 --- INVITE sip:12127773...@xxx.xxx.xxx.xxx SIP/2.0 To: sip:12127773...@xxx.xxx.xxx.xxx From: 100sip:1...@xxx.xxx.xxx.xxx;tag=c4446262 Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-422974952-1--d87543-;rport Call-ID: 052fcf17df558f7b CSeq: 1 INVITE Contact: sip:1...@201.xxx.xxx.xxx:5060 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Content-Length: 233 v=0 o=- 872302269 872302706 IN IP4 201.xxx.xxx.xxx s=eyeBeam c=IN IP4 201.xxx.xxx.xxx t=0 0 m=audio 8612 RTP/AVP 18 101 a=alt:1 1 : 88385B47 0038 201.xxx.xxx.xxx 8612 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv - [May 28 11:51:34] --- (12 headers 10 lines) --- [May 28 11:51:34] == Using SIP RTP CoS mark 5 [May 28 11:51:34] Sending to 201.xxx.xxx.xxx : 5060 (no NAT) [May 28 11:51:34] Using INVITE request as basis request - 052fcf17df558f7b [May 28 11:51:34] Found peer '100' for '100' from 201.xxx.xxx.xxx:5060 [0K[May 28 11:51:34] --- Reliably Transmitting (NAT) to 201.xxx.xxx.xxx:5060 --- SIP/2.0 401 Unauthorized v: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-422974952-1--d87543-;received=201.xxx.xxx.xxx;rport=5060 f: 100sip:1...@xxx.xxx.xxx.xxx;tag=c4446262 t: sip:12127773...@xxx.xxx.xxx.xxx;tag=as22c91f20 i: 052fcf17df558f7b CSeq: 1 INVITE Server: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO k: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm=asterisk, nonce=75cff649 l: 0 [May 28 11:51:34] Scheduling destruction of SIP dialog '052fcf17df558f7b' in 32000 ms (Method: INVITE) [May 28 11:51:34] --- SIP read from UDP:201.xxx.xxx.xxx:5060 --- ACK sip:12127773...@xxx.xxx.xxx.xxx SIP/2.0 To: sip:12127773...@xxx.xxx.xxx.xxx;tag=as22c91f20 From: 100sip:1...@xxx.xxx.xxx.xxx;tag=c4446262 Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-422974952-1--d87543-;rport Call-ID: 052fcf17df558f7b CSeq: 1 ACK Content-Length: 0 - [May 28 11:51:34] --- (7 headers 0 lines) --- [May 28 11:51:34] --- SIP read from UDP:201.xxx.xxx.xxx:5060 --- INVITE sip:12127773...@xxx.xxx.xxx.xxx SIP/2.0 To: sip:12127773...@xxx.xxx.xxx.xxx From: 100sip:1...@xxx.xxx.xxx.xxx;tag=c4446262 Via: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;rport Call-ID: 052fcf17df558f7b CSeq: 2 INVITE Contact: sip:1...@201.xxx.xxx.xxx:5060 Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3007n stamp 17816 Authorization: Digest username=100,realm=asterisk,nonce=75cff649,uri=sip:12127773...@xxx.xxx.xxx.xxx,response=ffeaf69c547dd3d1252e1bc7ab614fea,algorithm=MD5 Content-Length: 233 v=0 o=- 872302269 872302706 IN IP4 201.xxx.xxx.xxx s=eyeBeam c=IN IP4 201.xxx.xxx.xxx t=0 0 m=audio 8612 RTP/AVP 18 101 a=alt:1 1 : 88385B47 0038 201.xxx.xxx.xxx 8612 a=fmtp:101 0-15 a=rtpmap:101 telephone-event/8000 a=sendrecv - [May 28 11:51:34] --- (13 headers 10 lines) --- [May 28 11:51:34] Sending to 201.xxx.xxx.xxx : 5060 (NAT) [May 28 11:51:34] Using INVITE request as basis request - 052fcf17df558f7b [May 28 11:51:34] Found peer '100' for '100' from 201.xxx.xxx.xxx:5060 [May 28 11:51:34] Found RTP audio format 18 [May 28 11:51:34] Found RTP audio format 101 [May 28 11:51:34] Found audio description format telephone-event for ID 101 [May 28 11:51:34] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) [May 28 11:51:34] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [May 28 11:51:34] Peer audio RTP is at port 201.xxx.xxx.xxx:8612 [May 28 11:51:34] Looking for 12127773456 in asterisk (domain xxx.xxx.xxx.xxx) [May 28 11:51:34] list_route: hop: sip:1...@201.xxx.xxx.xxx:5060 [May 28 11:51:34] --- Transmitting (NAT) to 201.xxx.xxx.xxx:5060 --- SIP/2.0 100 Trying v: SIP/2.0/UDP 201.xxx.xxx.xxx:5060;branch=z9hG4bK-d87543-61806499-1--d87543-;received=201.xxx.xxx.xxx;rport=5060 f: 100sip:1...@xxx.xxx.xxx.xxx;tag=c4446262 t: sip:12127773...@xxx.xxx.xxx.xxx i: 052fcf17df558f7b CSeq: 2 INVITE Server: PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO k: replaces, timer m: sip:12127773...@xxx.xxx.xxx.xxx l: 0 [May 28 11:51:34] -- Executing AGI(SIP/100-115f, call.php) [May 28 11:51:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php [May 28 11:51:34] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456) [May 28 11:51:34] == Using SIP RTP CoS mark 5 [May 28 11:51:34] -- Couldn't call
Re: [asterisk-users] DTMF recognized after call establishment
So any resolution for this? I suspect it could be related to RE INVITE On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad asghar...@gmail.comwrote: i had this in past there was an ATA configured to send 9 at the end of dialing in my case. On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on SIP/MyTrunk-000a4b49, duration 0 ms [May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin '*' on SIP/MyTrunk-000a4b49 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on SIP/MyTrunk-000a4b49, duration 0 ms [May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin '8' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on SIP/MAN-000a4af0, duration 100 ms [May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8' with duration 100 queued on SIP/MAN-000a4af0 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8' queued on SIP/MAN-000a4af0 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on SIP/MAN-000a4b41, duration 100 ms [May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1' with duration 100 queued on SIP/MAN-000a4b41 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1' queued on SIP/MAN-000a4b41 [May 17 00:33:55] VERBOSE[4106] pbx.c: == Spawn extension (sip-trunk-inbound, 2127773456, 1) exited non-zero on 'SIP/MyTrunk-000a4af3' [May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing [h@trunk-outbound:1] NoOp(SIP/MAN-000a4b09, 16) in new stack [May 17 00:33:56] VERBOSE[4136] pbx.c: == Spawn extension (trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09' Is this some thing related to SIP RE-INVITE? Thanks. -- _ -- 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] DTMF recognized after call establishment
work around was block dtmf. set wrong type of dtmf in incoming trunk. On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: So any resolution for this? I suspect it could be related to RE INVITE On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad asghar...@gmail.comwrote: i had this in past there was an ATA configured to send 9 at the end of dialing in my case. On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on SIP/MyTrunk-000a4b49, duration 0 ms [May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin '*' on SIP/MyTrunk-000a4b49 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on SIP/MyTrunk-000a4b49, duration 0 ms [May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin '8' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on SIP/MAN-000a4af0, duration 100 ms [May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8' with duration 100 queued on SIP/MAN-000a4af0 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8' queued on SIP/MAN-000a4af0 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on SIP/MAN-000a4b41, duration 100 ms [May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1' with duration 100 queued on SIP/MAN-000a4b41 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1' queued on SIP/MAN-000a4b41 [May 17 00:33:55] VERBOSE[4106] pbx.c: == Spawn extension (sip-trunk-inbound, 2127773456, 1) exited non-zero on 'SIP/MyTrunk-000a4af3' [May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing [h@trunk-outbound:1] NoOp(SIP/MAN-000a4b09, 16) in new stack [May 17 00:33:56] VERBOSE[4136] pbx.c: == Spawn extension (trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09' Is this some thing related to SIP RE-INVITE? Thanks. -- _ -- 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 -- _ -- 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 11 dtmf not recognised
On Sat, May 25, 2013 at 10:32 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Finally got it working with 3 attempts by the fialplan,
exten = 300,1,Playback(letters/a)
exten = 300,n,Set(gottries=0)
exten = 300,n(getmore),Set(rightPIN=1)
exten = 300,n,Read(inPIN,,1,skip,3,3) ; Attempts for 3 times with 3
seconds of timeout
exten = 300,n(gotdigit),GotoIf($[${inPIN} =
${rightPIN}]?pin-accepted,1)
exten = 300,n,Set(gottries=$[${gottries}+1];
exten = 300,n,GotoIf($[${LEN(${inPIN})} == 0]?reallynothing:gotdigit)
exten = 300,n(reallynothing),GotoIf($[${gottries}3]?done:getmore) ;
Attempts for 3 tries if greater than 3 then it will come out or else
getmore will called
exten = 300,n(done),Playback(letters/c) ; Didn't go to pin-accepted, so
play badPIN and hangup
exten = pin-accepted,1,Playback(letters/b) ; correct pin, play
Thanks
On 25 May 2013 15:38, Gopalakrishnan N gopalakrishnan...@gmail.com
wrote:
Am using Read application to get the digit, since its recognizing... I
would like to get for 3 attempts and then after 3rd attempt it has to
playback some different message like entries exceeded.
My dialplan as,
exten = 100,1(begin),Playback(letters/a)
exten = 100,n,Set(rightPIN=1)
exten = 100,n,Read(inPIN,,1,skip,3,3) ; Attempts for 5 times with 3
seconds of timeout
exten = 100,n,GotoIf($[${inPIN} = ${rightPIN}]?pin-accepted,1)
exten = 100,n,Playback(letters/c) ; Didn't go to pin-accepted, so play
badPIN and hangup
exten = pin-accepted,1,Playback(letters/b) ; correct pin, play
what happens its keep on asking to enter digit If my DTMF didnt match. Do
i need to use any return function... ?
Actually my goal is to ask for 3 times and if not matched then return to
some other application.
Thanks in advance.
On Sat, May 25, 2013 at 3:19 PM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
With Asterisk 1.8 I got it working.
Regards
On Sat, May 25, 2013 at 2:37 AM, Gopalakrishnan N
gopalakrishnan...@gmail.com wrote:
Tried info, rfc2833, inband and finally kept as auto.
On 25 May 2013 02:20, Doug Lytle supp...@drdos.info wrote:
dtmfmode=auto
dtmfmode=info
or
dtmfmode=rfc2833
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little
Temporary Safety, deserve neither Liberty nor Safety.
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Syntax check:
exten = 300,n,Set(gottries=$[${gottries}+1];
should be:
exten = 300,n,Set(gottries=$[${gottries}+1])
--
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Re: [asterisk-users] RED on DAHDI channel
On Mon, May 27, 2013 at 12:14:41PM -0500, Mitch Claborn wrote: Asterisk 11.1 We have a situation where one of our incomings POTS lines will not answer. There are 2 lines configured by the Telco as a rollover group (rings the line that is not busy) and they feed into a Digium AEX410 on the server. The most recent time this happened, I did a /etc/init.d/dahdi status and saw this: ### Span 4: WCTDM/1 Wildcard AEX410 *53 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED* 54 FXOFXSKS (EC: VPMOCT032 - INACTIVE) 55 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 56 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 55 and 56 are always red - there is nothing plugged into those ports. 53 and 54 are the active lines. I restarted dahdi (/etc/init.d/dahdi stop then start) and it started working again, and the RED on 53 was gone. Is there something else I can do to try and figure out what is going on, and maybe how to prevent it? Hi Mitch, What version of DAHDI are you using? Unfortunately I did insert a bug on 2.6.0 where it was possible for a channel on an AEX410 to get stuck in RED alarm depending on the timing from the central office. If you're not using 2.6.0+ you can ignore the remainder of this email. The bug was fixed in 2.6.2 in wctdm24xxp: Eliminate chance for channel to be stuck in RED alarm. [1] but unfortunately, that fix had a problem of it's own which was fixed in wctdm24xxp: Fix FXO failure to detect battery CO disconnects. [2]. This just means there isn't currently a release of the 2.6 branch that contains all the recommended fixes. If you were on the 2.6 branch, then I advise installing the current tip of the 2.6.y branch like: $ git clone git://git.asterisk.org/dahdi/linux dahdi-linux -b 2.6.y $ cd dahdi-linux $ make install [1] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=41639330a59 [2] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=160edc8c9db If you don't have git installed on the machine you would like to install this on, you can use the 'snapshot' link when looking at the shortlog of the 2.6.y branch at git.asterisk.org [3] which will allow you to download a tar.gz file. [3] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/heads/2.6.y -- 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] Not able to build the chan_sip.c module
El 27/05/13 01:56, upendra escribió: Hi, i am trying to install asterisk newer version on the Elastix machine, but while installing the chan_sip,c module is not building while make. when i see in make menuselect options it showing XXX -- extended , please let me know how to enable it and make build chan_sip module. Why are you trying to build asterisk manually on the Elastix machine? Have you discussed your needs in the Elastix forums? If so required, you can try to tweak the asterisk source RPM from the Elastix project. -- _ -- 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 - Soundcard - Recording?
Greetings- I've got a curious project that I could use some input on. I'd like to use Asterisk to record some audio channels via USB 'soundcard'. When audio passes through the soundcard, Asterisk should grab that audio channel (CONSOLE?), and write it to a wav file. I'm perfectly competent with the dialplan portion of the recording, but I don't know about the following: -How does Asterisk know a new audio stream/source is beginning/ending? -Can I have more than one CONSOLE device -Is ALSA or OSS the preferred audio system with Asterisk these days (Asterisk 11 presumably) Any thoughts? Or, do you have any alternative ideas that would work better than using Asterisk for this? Thanks! --Tim -- _ -- 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 - Soundcard - Recording?
What are you trying to accomplish? What is the USB 'sound card' attached to? Your description is too cryptic for someone to propose a solution. Ron On 28/05/2013 12:45 PM, Tim Nelson wrote: Greetings- I've got a curious project that I could use some input on. I'd like to use Asterisk to record some audio channels via USB 'soundcard'. When audio passes through the soundcard, Asterisk should grab that audio channel (CONSOLE?), and write it to a wav file. I'm perfectly competent with the dialplan portion of the recording, but I don't know about the following: -How does Asterisk know a new audio stream/source is beginning/ending? -Can I have more than one CONSOLE device -Is ALSA or OSS the preferred audio system with Asterisk these days (Asterisk 11 presumably) Any thoughts? Or, do you have any alternative ideas that would work better than using Asterisk for this? Thanks! --Tim -- _ -- 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 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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 - Soundcard - Recording?
- Original Message - What are you trying to accomplish? What is the USB 'sound card' attached to? Your description is too cryptic for someone to propose a solution. The target use is to record mic level audio from various devices (could be an omnidirectional room mike, phone handset, etc). --Tim -- _ -- 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] RED on DAHDI channel
I am running 2.6.1. I'll give the 2.6.y a try. Mitch On 05/28/2013 10:53 AM, Shaun Ruffell wrote: On Mon, May 27, 2013 at 12:14:41PM -0500, Mitch Claborn wrote: Asterisk 11.1 We have a situation where one of our incomings POTS lines will not answer. There are 2 lines configured by the Telco as a rollover group (rings the line that is not busy) and they feed into a Digium AEX410 on the server. The most recent time this happened, I did a /etc/init.d/dahdi status and saw this: ### Span 4: WCTDM/1 Wildcard AEX410 *53 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED* 54 FXOFXSKS (EC: VPMOCT032 - INACTIVE) 55 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 56 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 55 and 56 are always red - there is nothing plugged into those ports. 53 and 54 are the active lines. I restarted dahdi (/etc/init.d/dahdi stop then start) and it started working again, and the RED on 53 was gone. Is there something else I can do to try and figure out what is going on, and maybe how to prevent it? Hi Mitch, What version of DAHDI are you using? Unfortunately I did insert a bug on 2.6.0 where it was possible for a channel on an AEX410 to get stuck in RED alarm depending on the timing from the central office. If you're not using 2.6.0+ you can ignore the remainder of this email. The bug was fixed in 2.6.2 in wctdm24xxp: Eliminate chance for channel to be stuck in RED alarm. [1] but unfortunately, that fix had a problem of it's own which was fixed in wctdm24xxp: Fix FXO failure to detect battery CO disconnects. [2]. This just means there isn't currently a release of the 2.6 branch that contains all the recommended fixes. If you were on the 2.6 branch, then I advise installing the current tip of the 2.6.y branch like: $ git clone git://git.asterisk.org/dahdi/linux dahdi-linux -b 2.6.y $ cd dahdi-linux $ make install [1] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=41639330a59 [2] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=160edc8c9db If you don't have git installed on the machine you would like to install this on, you can use the 'snapshot' link when looking at the shortlog of the 2.6.y branch at git.asterisk.org [3] which will allow you to download a tar.gz file. [3] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/heads/2.6.y -- _ -- 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] RED on DAHDI channel
I got the following warning during the build. Is it anything to worry about? WARNING: could not find /home/mclaborn/asterisk/dahdi-2.6.y-snapshot-20130528-linux-20a479b/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_64.o.cmd for /home/mclaborn/asterisk/dahdi-2.6.y-snapshot-20130528-linux-20a479b/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o Mitch On 05/28/2013 12:37 PM, Mitch Claborn wrote: I am running 2.6.1. I'll give the 2.6.y a try. Mitch On 05/28/2013 10:53 AM, Shaun Ruffell wrote: On Mon, May 27, 2013 at 12:14:41PM -0500, Mitch Claborn wrote: Asterisk 11.1 We have a situation where one of our incomings POTS lines will not answer. There are 2 lines configured by the Telco as a rollover group (rings the line that is not busy) and they feed into a Digium AEX410 on the server. The most recent time this happened, I did a /etc/init.d/dahdi status and saw this: ### Span 4: WCTDM/1 Wildcard AEX410 *53 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED* 54 FXOFXSKS (EC: VPMOCT032 - INACTIVE) 55 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 56 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 55 and 56 are always red - there is nothing plugged into those ports. 53 and 54 are the active lines. I restarted dahdi (/etc/init.d/dahdi stop then start) and it started working again, and the RED on 53 was gone. Is there something else I can do to try and figure out what is going on, and maybe how to prevent it? Hi Mitch, What version of DAHDI are you using? Unfortunately I did insert a bug on 2.6.0 where it was possible for a channel on an AEX410 to get stuck in RED alarm depending on the timing from the central office. If you're not using 2.6.0+ you can ignore the remainder of this email. The bug was fixed in 2.6.2 in wctdm24xxp: Eliminate chance for channel to be stuck in RED alarm. [1] but unfortunately, that fix had a problem of it's own which was fixed in wctdm24xxp: Fix FXO failure to detect battery CO disconnects. [2]. This just means there isn't currently a release of the 2.6 branch that contains all the recommended fixes. If you were on the 2.6 branch, then I advise installing the current tip of the 2.6.y branch like: $ git clone git://git.asterisk.org/dahdi/linux dahdi-linux -b 2.6.y $ cd dahdi-linux $ make install [1] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=41639330a59 [2] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=160edc8c9db If you don't have git installed on the machine you would like to install this on, you can use the 'snapshot' link when looking at the shortlog of the 2.6.y branch at git.asterisk.org [3] which will allow you to download a tar.gz file. [3] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/heads/2.6.y -- _ -- 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 - Soundcard - Recording?
On 28/05/2013 1:23 PM, Tim Nelson wrote: - Original Message - What are you trying to accomplish? What is the USB 'sound card' attached to? Your description is too cryptic for someone to propose a solution. The target use is to record mic level audio from various devices (could be an omnidirectional room mike, phone handset, etc). --Tim -- _ -- 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 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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 - Soundcard - Recording?
Sorry for the blank message. Fingers pressed send while brain was disenaged. Would Audacity be a better choice? http://wiki.audacityteam.org/wiki/Multichannel_Recording Ron On 28/05/2013 1:23 PM, Tim Nelson wrote: - Original Message - What are you trying to accomplish? What is the USB 'sound card' attached to? Your description is too cryptic for someone to propose a solution. The target use is to record mic level audio from various devices (could be an omnidirectional room mike, phone handset, etc). --Tim -- _ -- 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 -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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] Initial cut off audio
It seems that initial audio for SIP channels does not get transmitted for a period of varying length, typically about 1 second. This also applies to bridged SIP calls as well to one-legged calls where only Playback() gets called. The Definitive Asterisk Guide uses constructs like silence/1 or Wait() extensively and the explanation given in the text is to establish audio, if I remember this correctly. Normally, this delay does not seem to be a problem, but I have two installations (restaurants---because every syllable seems to be important when they shout at each other) that are problematic and where I got complaints about the initially cut off audio. Does somebody know whether the delay in establishing the audio signal is a typical Asterisk problem or are all VoIP solutions are affected? I am also not sure about the real cause. Is it really Asterisk that needs some time for the RTP streams or are the SIP phones responsible? jg -- _ -- 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 - Soundcard - Recording?
- Original Message - Sorry for the blank message. Fingers pressed send while brain was disenaged. Would Audacity be a better choice? http://wiki.audacityteam.org/wiki/Multichannel_Recording It would absolutely be a better solution. However, the recording is to be automated on a small system with no GUI, only console/SSH access. As such, running a full featured audio recording/mixing application in realtime (with user control) is not an option. :/ --Tim -- _ -- 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] RED on DAHDI channel
On Tue, May 28, 2013 at 12:44:47PM -0500, Mitch Claborn wrote: I got the following warning during the build. Is it anything to worry about? WARNING: could not find /home/mclaborn/asterisk/dahdi-2.6.y-snapshot-20130528-linux-20a479b/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_64.o.cmd for /home/mclaborn/asterisk/dahdi-2.6.y-snapshot-20130528-linux-20a479b/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o No, that's normal. It's a side effect that the compiler doesn't know all the options that were used to produce the precompile VPMADT032 loader. -- 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 - Soundcard - Recording?
You are still being a bit evasive but should I understand that you want to run a headless machine with open microphones that records what ever it hears? What do you want to do with each sound bite? How long does the silence have to be before you close the recording and dispose of it (save, e-mail, upload, whatever). Sounds like a security monitoring package (minus the video) should do the job? A little Googleing shows up these. http://oreka.sourceforge.net/about/ http://www.ubuntugeek.com/how-to-recording-internal-audio-in-ubuntu.html What else do you want it to do? Ron On 28/05/2013 2:23 PM, Tim Nelson wrote: - Original Message - Sorry for the blank message. Fingers pressed send while brain was disenaged. Would Audacity be a better choice? http://wiki.audacityteam.org/wiki/Multichannel_Recording It would absolutely be a better solution. However, the recording is to be automated on a small system with no GUI, only console/SSH access. As such, running a full featured audio recording/mixing application in realtime (with user control) is not an option. :/ --Tim -- Ron Wheeler President Artifact Software Inc email: rwhee...@artifact-software.com skype: ronaldmwheeler phone: 866-970-2435, ext 102 -- _ -- 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 - Soundcard - Recording?
I'll take a stab, since you said no GUI and also USB based mic. Raspberry Pi project? I'm interested in this vein as well, especially after the recent post about voice recognition. I was thinking that Raspberry Pi's with mics could live around my house and all have dedicated always-open channels to a conference bridge in the main asterisk box. I was planning on using ALSA and a USB mic on a local Raspberry Pi asterisk instance. So given that we know basically what you are trying to do, the original question was OSS versus ALSA for USB mic, correct? Has anyone had any thoughts on that? I thought ALSA was built in to the kernel and OSS required some hacks. But that is a pretty fuzzy recollection. j On 05/28/2013 03:10 PM, Ron Wheeler wrote: You are still being a bit evasive but should I understand that you want to run a headless machine with open microphones that records what ever it hears? What do you want to do with each sound bite? How long does the silence have to be before you close the recording and dispose of it (save, e-mail, upload, whatever). Sounds like a security monitoring package (minus the video) should do the job? A little Googleing shows up these. http://oreka.sourceforge.net/about/ http://www.ubuntugeek.com/how-to-recording-internal-audio-in-ubuntu.html What else do you want it to do? Ron On 28/05/2013 2:23 PM, Tim Nelson wrote: - Original Message - Sorry for the blank message. Fingers pressed send while brain was disenaged. Would Audacity be a better choice? http://wiki.audacityteam.org/wiki/Multichannel_Recording It would absolutely be a better solution. However, the recording is to be automated on a small system with no GUI, only console/SSH access. As such, running a full featured audio recording/mixing application in realtime (with user control) is not an option. :/ --Tim -- _ -- 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 - Soundcard - Recording?
So what is the fix for it? It was working all fine, all of a sudden it stopped working. Regards Jitesh Gala Director Fantasia Business Park, Nano Wing, S-10, Sector-30A, Vashi, Navi Mumbai - 400705 Cell No: +91 9769144905 Skype ID: jitesh_gala www.hubrisbpo.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff LaCoursiere Sent: Wednesday, May 29, 2013 1:50 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk - Soundcard - Recording? I'll take a stab, since you said no GUI and also USB based mic. Raspberry Pi project? I'm interested in this vein as well, especially after the recent post about voice recognition. I was thinking that Raspberry Pi's with mics could live around my house and all have dedicated always-open channels to a conference bridge in the main asterisk box. I was planning on using ALSA and a USB mic on a local Raspberry Pi asterisk instance. So given that we know basically what you are trying to do, the original question was OSS versus ALSA for USB mic, correct? Has anyone had any thoughts on that? I thought ALSA was built in to the kernel and OSS required some hacks. But that is a pretty fuzzy recollection. j On 05/28/2013 03:10 PM, Ron Wheeler wrote: You are still being a bit evasive but should I understand that you want to run a headless machine with open microphones that records what ever it hears? What do you want to do with each sound bite? How long does the silence have to be before you close the recording and dispose of it (save, e-mail, upload, whatever). Sounds like a security monitoring package (minus the video) should do the job? A little Googleing shows up these. http://oreka.sourceforge.net/about/ http://www.ubuntugeek.com/how-to-recording-internal-audio-in-ubuntu.html What else do you want it to do? Ron On 28/05/2013 2:23 PM, Tim Nelson wrote: - Original Message - Sorry for the blank message. Fingers pressed send while brain was disenaged. Would Audacity be a better choice? http://wiki.audacityteam.org/wiki/Multichannel_Recording It would absolutely be a better solution. However, the recording is to be automated on a small system with no GUI, only console/SSH access. As such, running a full featured audio recording/mixing application in realtime (with user control) is not an option. :/ --Tim -- _ -- 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] RED on DAHDI channel
Wednesday AM I hope Connected by DROID on Verizon Wireless -Original message- From: Shaun Ruffell sruff...@digium.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tue, May 28, 2013 15:53:45 GMT+00:00 Subject: Re: [asterisk-users] RED on DAHDI channel On Mon, May 27, 2013 at 12:14:41PM -0500, Mitch Claborn wrote: Asterisk 11.1 We have a situation where one of our incomings POTS lines will not answer. There are 2 lines configured by the Telco as a rollover group (rings the line that is not busy) and they feed into a Digium AEX410 on the server. The most recent time this happened, I did a /etc/init.d/dahdi status and saw this: ### Span 4: WCTDM/1 Wildcard AEX410 *53 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED* 54 FXOFXSKS (EC: VPMOCT032 - INACTIVE) 55 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 56 FXOFXSKS (EC: VPMOCT032 - INACTIVE) RED 55 and 56 are always red - there is nothing plugged into those ports. 53 and 54 are the active lines. I restarted dahdi (/etc/init.d/dahdi stop then start) and it started working again, and the RED on 53 was gone. Is there something else I can do to try and figure out what is going on, and maybe how to prevent it? Hi Mitch, What version of DAHDI are you using? Unfortunately I did insert a bug on 2.6.0 where it was possible for a channel on an AEX410 to get stuck in RED alarm depending on the timing from the central office. If you're not using 2.6.0+ you can ignore the remainder of this email. The bug was fixed in 2.6.2 in wctdm24xxp: Eliminate chance for channel to be stuck in RED alarm. [1] but unfortunately, that fix had a problem of it's own which was fixed in wctdm24xxp: Fix FXO failure to detect battery CO disconnects. [2]. This just means there isn't currently a release of the 2.6 branch that contains all the recommended fixes. If you were on the 2.6 branch, then I advise installing the current tip of the 2.6.y branch like: $ git clone git://git.asterisk.org/dahdi/linux dahdi-linux -b 2.6.y $ cd dahdi-linux $ make install [1] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=41639330a59 [2] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=commit;h=160edc8c9db If you don't have git installed on the machine you would like to install this on, you can use the 'snapshot' link when looking at the shortlog of the 2.6.y branch at git.asterisk.org [3] which will allow you to download a tar.gz file. [3] http://git.asterisk.org/gitweb/?p=dahdi/linux.git;a=shortlog;h=refs/heads/2.6 .y -- 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 -- _ -- 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] RED on DAHDI channel
The 2.6.y version installed without issue. A few test calls went OK. Will leave it in and see how things go. The problem has been sporadic, so won't know for a while if the issue is solved. Mitch On 05/28/2013 01:37 PM, Shaun Ruffell wrote: On Tue, May 28, 2013 at 12:44:47PM -0500, Mitch Claborn wrote: I got the following warning during the build. Is it anything to worry about? WARNING: could not find /home/mclaborn/asterisk/dahdi-2.6.y-snapshot-20130528-linux-20a479b/drivers/dahdi/vpmadt032_loader/.vpmadt032_x86_64.o.cmd for /home/mclaborn/asterisk/dahdi-2.6.y-snapshot-20130528-linux-20a479b/drivers/dahdi/vpmadt032_loader/vpmadt032_x86_64.o No, that's normal. It's a side effect that the compiler doesn't know all the options that were used to produce the precompile VPMADT032 loader. -- _ -- 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] G.729 codec in pass-thru mode
Kamlesh, Please provide SIP traces of both call legs for a failed call. Your last message only included a SIP trace of the call leg from the SIP softphone to the Asterisk server. There was no SIP trace for the call leg from the Asterisk server to the ITSP and, as shown below, that is probably where the answer to your problem can be found. First, the call leg from the SIP softphone to the Asterisk server successfully negotiated G.729 as the codec: [May 28 11:51:34] Found RTP audio format 18 ... [May 28 11:51:34] Capabilities: us - 0x104 (ulaw|g729), peer - audio=0x100 (g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x100 (g729) However, the call.php AGI script then failed to create the call leg from the Asterisk server to the ITSP: [May 28 11:51:34] -- Executing AGI(SIP/100-115f, call.php) [May 28 11:51:34] -- Launched AGI Script /var/lib/asterisk/agi-bin/call.php [May 28 11:51:34] -- AGI Script Executing Application: (Dial) Options: (SIP/yyy.yyy.yyy.yyy/12127773456) [May 28 11:51:34] == Using SIP RTP CoS mark 5 [May 28 11:51:34] -- Couldn't call yyy.yyy.yyy.yyy/12127773456 [May 28 11:51:34] Scheduling destruction of SIP dialog '142182ef20750fda512f8d2b0b071...@xxx.xxx.xxx.xxx' in 32000 ms (Method: INVITE) [May 28 11:51:34] == Everyone is busy/congested at this time (0:0/0/0) [May 28 11:51:34] -- SIP/100-115fAGI Script call.php completed, returning 0 [May 28 11:51:34] -- Auto fallthrough, channel 'SIP/100-115f' status is 'CHANUNAVAIL' Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- 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] DTMF recognized after call establishment
Let me try with dtmfmode as auto... On 28 May 2013 19:32, Asghar Mohammad asghar...@gmail.com wrote: work around was block dtmf. set wrong type of dtmf in incoming trunk. On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: So any resolution for this? I suspect it could be related to RE INVITE On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad asghar...@gmail.comwrote: i had this in past there was an ATA configured to send 9 at the end of dialing in my case. On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end '*' received on SIP/MyTrunk-000a4b49, duration 0 ms [May 17 00:33:35] DTMF[4238] channel.c: DTMF end accepted without begin '*' on SIP/MyTrunk-000a4b49 [May 17 00:33:35] DTMF[4238] channel.c: DTMF end passthrough '*' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end '8' received on SIP/MyTrunk-000a4b49, duration 0 ms [May 17 00:33:36] DTMF[4238] channel.c: DTMF end accepted without begin '8' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4238] channel.c: DTMF end passthrough '8' on SIP/MyTrunk-000a4b49 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end '8' received on SIP/MAN-000a4af0, duration 100 ms [May 17 00:33:36] DTMF[4104] channel.c: DTMF begin emulation of '8' with duration 100 queued on SIP/MAN-000a4af0 [May 17 00:33:36] DTMF[4104] channel.c: DTMF end emulation of '8' queued on SIP/MAN-000a4af0 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end '1' received on SIP/MAN-000a4b41, duration 100 ms [May 17 00:33:37] DTMF[4234] channel.c: DTMF begin emulation of '1' with duration 100 queued on SIP/MAN-000a4b41 [May 17 00:33:37] DTMF[4234] channel.c: DTMF end emulation of '1' queued on SIP/MAN-000a4b41 [May 17 00:33:55] VERBOSE[4106] pbx.c: == Spawn extension (sip-trunk-inbound, 2127773456, 1) exited non-zero on 'SIP/MyTrunk-000a4af3' [May 17 00:33:56] VERBOSE[4136] pbx.c: -- Executing [h@trunk-outbound:1] NoOp(SIP/MAN-000a4b09, 16) in new stack [May 17 00:33:56] VERBOSE[4136] pbx.c: == Spawn extension (trunk-outbound, 87457712, 2) exited non-zero on 'SIP/MAN-000a4b09' Is this some thing related to SIP RE-INVITE? Thanks. -- _ -- 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 -- _ -- 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
