[asterisk-users] FW: question on how to connect 2 boxes
Was my question not understood? Hello, I would like to connect 2 asterisk boxes together, so this is my scenario: Asterisk Main: it is connected to many sip providers and its main purpose as a call termination forwarder. Asterisk B: its connected to E1, and its purpose to terminate calls. It will receive SIP messages from Asterisk_Main, but there will be no voice traffic going between them, Asterisk_Main will send the provider IP address where both Asterisk_B provider will communicate, at the end of the call, Asterisk_Main will log CDR traffic. Now I can make IAX2, but the problem is, the traffic must go like this: Provider -Asterisk_Main- Asterisk_B This will cause a problem with wasted bandwidth and more latency on the call. I want it to be like this: Asterisk_Main | Asterisk_B -ßà Provider Please let me know your suggestions. Regards. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] question on how to connect 2 boxes
Hello, I would like to connect 2 asterisk boxes together, so this is my scenario: Asterisk Main: it is connected to many sip providers and its main purpose as a call termination forwarder. Asterisk B: its connected to E1, and its purpose to terminate calls. It will receive SIP messages from Asterisk_Main, but there will be no voice traffic going between them, Asterisk_Main will send the provider IP address where both Asterisk_B provider will communicate, at the end of the call, Asterisk_Main will log CDR traffic. Now I can make IAX2, but the problem is, the traffic must go like this: Provider -Asterisk_Main- Asterisk_B This will cause a problem with wasted bandwidth and more latency on the call. I want it to be like this: Asterisk_Main | Asterisk_B -ßà Provider Please let me know your suggestions. Regards. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need help with this conf
Hello, I would appreciate if someone can give some help on what I want: When someone call my box (from outside), to a certain ZAP port, it will put him on hold, and immediately the box calls to outside SIP trunk to a preconfigured certain number, then when the other party picks up the phone, both calls connected and CDR starts counting. Any idea? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IVR for asterisk
Anyone can recommend a commercial large scale IVR with easy + pro management for asterisk? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] max call duration --- SOLVED
Thanks, I just added this: exten = s,n,Set(TIMEOUT(absolute)=360) will limit 3 minutes (including dialing) -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Barry L. Kline Sent: Tuesday, November 17, 2009 4:50 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] max call duration -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 B.Masoud @ SH wrote: How can I set a maximum call duration on a ZAP channel? Look at the parameters on the Dial application. Barry -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.5 (GNU/Linux) iD8DBQFLAqoTCFu3bIiwtTARAiULAJ9E3g1x5lY5yspsXVKgz3yAFFOAqgCfV9Fy GnifFRJRrv98EWIgzK+RvKw= =UT+T -END PGP SIGNATURE- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] max call duration
How can I set a maximum call duration on a ZAP channel? Thank you. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk SIP hangup
Hello all, How can I ask Asterisk to ignore a sip hang-up request for XX seconds from the beginning of the session? Thank you ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Termination Question
Hello, I would like to know how the following scenario works: I have 3 Asterisk servers, A,B C, each one is located in a different country. Asterisk A is the main one, and both B C are connected to it. My question is, when a call is originated from B to C, it will have to go through A, but does A makes a peer connection between B C to eliminate bandwidth and latency, or the call has to go through A ??? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Termination Question
So how can I let A makes a PEER connection between B C, and ONLY log the call information? Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Thursday, November 12, 2009 6:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Termination Question ...and with a packet switched transport layer, the 'hairpin' route through A may create problematic levels of latency--latency that would perhaps NOT have been problematic on a classic circuit switched route, so it's definitely advisable to nail up a connection between b and c. -K - Original Message - From: Tarek Sawah mailto:tareksa...@hotmail.com To: Asterisk Users mailto:asterisk-users@lists.digium.com Sent: Thursday, November 12, 2009 8:28 AM Subject: Re: [asterisk-users] Termination Question for the sake of bandwidth you are supposed to connect each two servers together.. otherwise calls between B C will have to go through A . -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ From: i...@saudihome.com To: asterisk-users@lists.digium.com Date: Thu, 12 Nov 2009 16:13:10 +0300 Subject: [asterisk-users] Termination Question Hello, I would like to know how the following scenario works: I have 3 Asterisk servers, A,B C, each one is located in a different country. Asterisk A is the main one, and both B C are connected to it. My question is, when a call is originated from B to C, it will have to go through A, but does A makes a peer connection between B C to eliminate bandwidth and latency, or the call has to go through A ??? Thanks. _ Windows 7: Unclutter your desktop. Learn more. http://go.microsoft.com/?linkid=9690331ocid=PID24727::T:WLMTAGL:ON:WL:en-U S:WWL_WIN_evergreen:112009 _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Termination Question
That could work, but I have no control over server B, not server C ! From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Friday, November 13, 2009 3:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Termination Question I have no first-hand experience with the fussy idiosyncrasies, but the BIG PICTURE is to have server A set up the call, and then reinvite the media directly from B to C. The call control messages flow to server A, the media goes directly. If you don't have NAT traversal Kung-Fu, I suggest using IAX2 over SIP. -K - Original Message - From: B.Masoud @ SH mailto:i...@saudihome.com To: 'Asterisk Users Mailing List - mailto:asterisk-users@lists.digium.com Non-Commercial Discussion' Sent: Thursday, November 12, 2009 6:10 PM Subject: Re: [asterisk-users] Termination Question So how can I let A makes a PEER connection between B C, and ONLY log the call information? Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Karl Fife Sent: Thursday, November 12, 2009 6:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Termination Question ...and with a packet switched transport layer, the 'hairpin' route through A may create problematic levels of latency--latency that would perhaps NOT have been problematic on a classic circuit switched route, so it's definitely advisable to nail up a connection between b and c. -K - Original Message - From: Tarek Sawah mailto:tareksa...@hotmail.com To: Asterisk Users mailto:asterisk-users@lists.digium.com Sent: Thursday, November 12, 2009 8:28 AM Subject: Re: [asterisk-users] Termination Question for the sake of bandwidth you are supposed to connect each two servers together.. otherwise calls between B C will have to go through A . -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 _ From: i...@saudihome.com To: asterisk-users@lists.digium.com Date: Thu, 12 Nov 2009 16:13:10 +0300 Subject: [asterisk-users] Termination Question Hello, I would like to know how the following scenario works: I have 3 Asterisk servers, A,B C, each one is located in a different country. Asterisk A is the main one, and both B C are connected to it. My question is, when a call is originated from B to C, it will have to go through A, but does A makes a peer connection between B C to eliminate bandwidth and latency, or the call has to go through A ??? Thanks. _ Windows 7: Unclutter your desktop. Learn more. http://go.microsoft.com/?linkid=9690331ocid=PID24727::T:WLMTAGL:ON:WL:en-U S:WWL_WIN_evergreen:112009 _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] outbound routing
I have 2 questions: 1. Can I make outbound route rule based on the Source Channel? 2. Can I auto change the outbound route based on time/Day of week? Any help very appreciated.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound routing
Can you tell me how on the first question? Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Alex Balashov Sent: Sunday, November 08, 2009 10:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] outbound routing -- Sent from mobile device On Nov 8, 2009, at 2:13 PM, B.Masoud @ SH i...@saudihome.com wrote: I have 2 questions: 1. Can I make outbound route rule based on the Source Channel? Yes. 2. Can I auto change the outbound route based on time/Day of week? Yes. See GotoIfTime(). Any help very appreciated.. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic DNS trunk --- SOLVED
dnsmgr.conf: enable=yes refreshinterval=300 regards. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 30, 2009 3:28 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Dynamic DNS trunk Thanks I did this dnsmgr.conf: enable=yes refreshinterval=300 I did dnsmgr refresh, the DNS in the trunk did not got the new ip, also I waited 5 min. do I have to add an entry to dnsmgr?? Thanks! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Friday, October 30, 2009 1:53 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dynamic DNS trunk On 30/10/09 6:42 AM, B.Masoud @ SH wrote: Hi I tried with registration, it did not update the IP address I can only see it updated if I typed: Sip reload I have few questions: Is there any way Asterisk automatically updates the DNS? Yep /etc/asterisk/dnsmgr.conf -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calls disconnects after short time
Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI -- Hungup 'IAX2/9-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/213.165.32.100-b7d21018' in macro 'dialout-trunk' == Spawn extension (outbound-allroutes, 966507944491, 4) exited non-zero on 'SIP/213.165.32.100-b7d21018' -- Executing [...@macro-dialout-trunk:1] Macro(SIP/213.165.32.100-b7d21018, hangupcall|) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/213.165.32.100-b7d21018, w) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/213.165.32.100-b7d21018, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/213.165.32.100-b7d21018, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/213.165.32.100-b7d21018, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/213.165.32.100-b7d21018, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/213.165.32.100-b7d21018, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/213.165.32.100-b7d21018' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/213.165.32.100-b7d21018' elastix*CLI ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls disconnects after short time
My server use public ip, so no nat issues, here is the out of sip debug: - --- (10 headers 0 lines) --- Sending to 213.165.32.100 : 5060 (no NAT) --- Reliably Transmitting (no NAT) to 213.165.32.100:5060 --- SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 213.165.32.100:5060;branch=z9hG4bKa95f47a28fdc61714dc862cefe1a326a;received= 213.165.32.100 From: sip:9991...@213.165.32.100;tag=3466008105-77358 To: 966599740196 sip:966599740...@213.165.32.100;tag=as54d7ac3d Call-ID: 19751463-3466008105-77...@dalmsx01.vincomm.net CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 elastix*CLI --- Transmitting (no NAT) to 213.165.32.100:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 213.165.32.100:5060;branch=z9hG4bKa95f47a28fdc61714dc862cefe1a326a;received= 213.165.32.100 From: sip:9991...@213.165.32.100;tag=3466008105-77358 To: 966599740196 sip:966599740...@213.165.32.100;tag=as54d7ac3d Call-ID: 19751463-3466008105-77...@dalmsx01.vincomm.net CSeq: 1 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 -- Hungup 'IAX2/9-4490' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' in macro 'dialout-trunk' == Spawn extension (outbound-allroutes, 966599740196, 4) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' -- Executing [...@macro-dialout-trunk:1] Macro(SIP/213.165.32.100-b7c10ad8, hangupcall|) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/213.165.32.100-b7c10ad8, w) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/213.165.32.100-b7c10ad8, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/213.165.32.100-b7c10ad8, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/213.165.32.100-b7c10ad8, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/213.165.32.100-b7c10ad8, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/213.165.32.100-b7c10ad8, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' in macro 'hangupcall' == Spawn extension (macro-dialout-trunk, h, 1) exited non-zero on 'SIP/213.165.32.100-b7c10ad8' elastix*CLI thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Sunday, November 01, 2009 1:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calls disconnects after short time Where is the log for the actual hang up of the call?.. can you do a sip debug? Although there can be many reasons, my first suspect is always a nat issue, which manifest as the inability of asterisk to receive the incoming packets. In that case, you should be getting a message saying hanging up call , no reply to our critical package. see if you receive a message like that in your debugging. CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Saturday, October 31, 2009 8:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Calls disconnects after short time Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? elastix*CLI -- Hungup 'IAX2/9-6813' == Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/213.165.32.100-b7d21018' in macro 'dialout-trunk' == Spawn extension (outbound-allroutes, 966507944491, 4) exited non-zero on 'SIP/213.165.32.100-b7d21018' -- Executing [...@macro-dialout-trunk:1] Macro(SIP/213.165.32.100-b7d21018, hangupcall|) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/213.165.32.100-b7d21018, w) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/213.165.32.100-b7d21018, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/213.165.32.100-b7d21018, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/213.165.32.100-b7d21018, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/213.165.32.100-b7d21018, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/213.165.32.100-b7d21018, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP
Re: [asterisk-users] Calls disconnects after short time
Hello, I have grabbed again a whole call when it hangs up debug, I dono what else I can read?? What exactly you want me to look for? And assuming there is a firewall at my ISP, how to diagnose it? Thanks for the advise, Here is another log: -- Called 9/0557202919 -- Call accepted by xxx.xxx.xxx.xxx (format ulaw) -- Format for call is ulaw elastix*CLI --- SIP read from xx.xx.xx.xx:5060 --- CANCEL sip:966557202...@xx.xx.xx.xx SIP/2.0 Max-Forwards: 70 To: 966557202919 sip:966557202...@xx.xx.xx.xx From: sip:9998...@xx.xx.xx.xx;tag=3466014864-147468 Call-ID: 19773310-3466014864-147...@aaa.bbb.net CSeq: 1 CANCEL Allow: INVITE, BYE, OPTIONS, CANCEL, ACK, REGISTER, NOTIFY, INFO, REFER, SUBSCRIBE, PRACK, UPDATE Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bKd6a812de72f094987d1363161c853aec Contact: sip:9998...@xx.xx.xx.xx:5060 Content-Length: 0 - --- (10 headers 0 lines) --- Sending to xx.xx.xx.xx : 5060 (no NAT) --- Reliably Transmitting (no NAT) to xx.xx.xx.xx:5060 --- SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bKd6a812de72f094987d1363161c853aec;received=xx. xx.xx.xx From: sip:9998...@xx.xx.xx.xx;tag=3466014864-147468 To: 966557202919 sip:966557202...@xx.xx.xx.xx;tag=as717c0994 Call-ID: 19773310-3466014864-147460@ aa.bb.net CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- Transmitting (no NAT) to xx.xx.xx.xx:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP xx.xx.xx.xx:5060;branch=z9hG4bKd6a812de72f094987d1363161c853aec;received=xx. xx.xx.xx From: sip:9998...@xx.xx.xx.xx;tag=3466014864-147468 To: 966557202919 sip:966557202...@xx.xx.xx.xx;tag=as717c0994 Call-ID: 19773310-3466014864-147...@aa.bb.net CSeq: 1 CANCEL User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 -- Hungup 'IAX2/9-8610' Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Sunday, November 01, 2009 4:11 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calls disconnects after short time The only informative part are the 2 paragraphs of the sip debug, but can't tell much since you only show a very small portion of the sip log. There is a 487 Request terminated there screaming at you but can't tell if meaning that provider is not handling the ACKs. That section of the [macro-hangupcall] context is useless as it is caused by the hangup, and not an effect. The usage of a public IP is not indicative of the existence of a firewall which can be blocking any necessary ports for tcp and/or udp. You should always cover your real IP numbers when showing samples of your logs CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Sunday, November 01, 2009 12:37 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calls disconnects after short time My server use public ip, so no nat issues, here is the out of sip debug: thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of C. Savinovich Sent: Sunday, November 01, 2009 1:18 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Calls disconnects after short time Where is the log for the actual hang up of the call?.. can you do a sip debug? Although there can be many reasons, my first suspect is always a nat issue, which manifest as the inability of asterisk to receive the incoming packets. In that case, you should be getting a message saying hanging up call , no reply to our critical package. see if you receive a message like that in your debugging. CS From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Saturday, October 31, 2009 8:12 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Calls disconnects after short time Hello, My client customers complaining that their calls suddenly get hung-up, I am just investigating if the problem from my side, I had a log of a hang-up case, Does it help to know if there is a problem that can be resolved from my side? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] strange dialing HELP !
Hello I just found out this: I had a phone into the FXO ports to see why calls are not passing through, When I ask asterisk to dial a number of 10 digits, it dials the first 9 digits, then wait 2 seconds and dial the last digit! Any idea how to overcome this and dial the whole number 1 shot The card I am using is TDM digium card 24 ports FXO. Thanks a lot! ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic DNS trunk
Hi I tried with registration, it did not update the IP address I can only see it updated if I typed: Sip reload I have few questions: Is there any way Asterisk automatically updates the DNS? If no other way, can I type sip reload on a production system safely? If yes, any help shows how to send the command sip reload periodically to asterisk? Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Juan E. Rodríguez Sent: Thursday, October 29, 2009 6:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dynamic DNS trunk If the trunk is a dynamic IP you need the other end to register to Asterisk, so letting Asterisk know the new IP. Regards, Juan B.Masoud @ SH wrote: I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dynamic DNS trunk
Thanks I did this dnsmgr.conf: enable=yes refreshinterval=300 I did dnsmgr refresh, the DNS in the trunk did not got the new ip, also I waited 5 min. do I have to add an entry to dnsmgr?? Thanks! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Friday, October 30, 2009 1:53 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Dynamic DNS trunk On 30/10/09 6:42 AM, B.Masoud @ SH wrote: Hi I tried with registration, it did not update the IP address I can only see it updated if I typed: Sip reload I have few questions: Is there any way Asterisk automatically updates the DNS? Yep /etc/asterisk/dnsmgr.conf -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dynamic DNS trunk
I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] hangup from which side
When Asterisk establish a call through an outbound trunk, Is there any way I can know who hang up the call first? The caller or the party called? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] polarity on some channels
Hello, I have : answeronpolarityswitch=yes on chan_dahdi.conf but it's making all my lines answer on polarity reversal, this causes a problem for PSTN lines, so how can I set these lines to answer immediately (when it rings)? thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polarity on some channels
It's not caller ID issue, I can make asterisk answer the line by omitting the line answeronpolarityswitch=no , but this will take effect on all 24 TDM channels, I want some to have answer on polarity, and some without polarity. Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Lyle Giese Sent: Wednesday, October 21, 2009 10:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] polarity on some channels B.Masoud @ SH wrote: Hello, I have : answeronpolarityswitch=yes on chan_dahdi.conf but it's making all my lines answer on polarity reversal, this causes a problem for PSTN lines, so how can I set these lines to answer immediately (when it rings)? thanks _ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Try turning off callerid. The 'standard' for POTS lines in the US is to put the caller id in between ring1 ring2. Asterisk waits for callerid before answering the line by default. usecallerid=off ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 1751 setup with asterisk
I have tried more than 10 different branded/non branded, audiocodes was by far the best fxo device.. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph Sent: Wednesday, October 21, 2009 12:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 1751 setup with asterisk On 10/20/09 09:01, Jonathan Thurman wrote: Not likely. Cisco works great with CallManager, but seems to be somewhat broken with anything else... wonder why? If you want something that is dependable and easy to configure I have had great success with the AudioCodes MP-114 devices. -Jonathan AudioCodes MP-114 is a bit out of my price range especially with echo cancellation module. But I just spotted: Sangoma's USBfxo unit at about $135.00 for two FXO is a reasonable deal. I wasn't able to find any reviews how it works with asterisk. -- Joseph ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] all our circuits are busy now
) in new stack -- Executing [...@macro-dialout-trunk:14] Set(IAX2/9-16336, custom=DAHDI/r1) in new stack -- Executing [...@macro-dialout-trunk:15] ExecIf(IAX2/9-16336, 0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)) in new stack -- Executing [...@macro-dialout-trunk:16] Macro(IAX2/9-16336, dialout-trunk-predial-hook|) in new stack -- Executing [...@macro-dialout-trunk-predial-hook:1] MacroExit(IAX2/9-16336, ) in new stack -- Executing [...@macro-dialout-trunk:17] GotoIf(IAX2/9-16336, 0?bypass|1) in new stack -- Executing [...@macro-dialout-trunk:18] GotoIf(IAX2/9-16336, 0?customtrunk) in new stack -- Executing [...@macro-dialout-trunk:19] Dial(IAX2/9-16336, DAHDI/r1/0505103250|300|) in new stack == Everyone is busy/congested at this time (1:0/1/0) -- Executing [...@macro-dialout-trunk:20] Goto(IAX2/9-16336, s-CONGESTION|1) in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,1) -- Executing [s-congest...@macro-dialout-trunk:1] GotoIf(IAX2/9-16336, 1?noreport) in new stack -- Goto (macro-dialout-trunk,s-CONGESTION,3) -- Executing [s-congest...@macro-dialout-trunk:3] NoOp(IAX2/9-16336, TRUNK Dial failed due to CONGESTION - failing through to other trunks) in new stack -- Executing [0505103...@from-internal:5] Macro(IAX2/9-16336, outisbusy|) in new stack -- Executing [...@macro-outisbusy:1] Playback(IAX2/9-16336, all-circuits-busy-now|noanswer) in new stack -- IAX2/9-16336 Playing 'all-circuits-busy-now' (language 'en') -- Executing [...@macro-outisbusy:2] Playback(IAX2/9-16336, pls-try-call-later|noanswer) in new stack -- IAX2/9-16336 Playing 'pls-try-call-later' (language 'en') -- Executing [...@macro-outisbusy:3] Macro(IAX2/9-16336, hangupcall) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(IAX2/9-16336, w) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(IAX2/9-16336, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(IAX2/9-16336, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(IAX2/9-16336, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(IAX2/9-16336, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(IAX2/9-16336, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'IAX2/9-16336' in macro 'hangupcall' == Spawn extension (macro-outisbusy, s, 3) exited non-zero on 'IAX2/9-16336' in macro 'outisbusy' == Spawn extension (from-internal, 0505103250, 5) exited non-zero on 'IAX2/9-16336' -- Executing [...@from-internal:1] Macro(IAX2/9-16336, hangupcall) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(IAX2/9-16336, w) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(IAX2/9-16336, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(IAX2/9-16336, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(IAX2/9-16336, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(IAX2/9-16336, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(IAX2/9-16336, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'IAX2/9-16336' in macro 'hangupcall' == Spawn extension (from-internal, s, 1) exited non-zero on 'IAX2/9-16336' -- Hungup 'IAX2/9-16336' -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Tuesday, October 20, 2009 4:35 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] all our circuits are busy now On 20/10/09 1:30 PM, B.Masoud @ SH wrote: I am not sure why I am getting this message, I have an outbound route that goes to asterisk gateway1 then asterisk gateway2 When all lines on asterisk gateway1 are full, I get the message all our circuits are busy now then few second later, the phone rings, going to the second route! And the call can be established, how can I get rid of this message?? On Asterisk 2 set the group to outbound lines or something, then check the number of channels in that group before making a call - if it's more than you have lines then respond with busy or something. -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list
[asterisk-users] all our circuits are busy now
I am not sure why I am getting this message, I have an outbound route that goes to asterisk gateway1 then asterisk gateway2 When all lines on asterisk gateway1 are full, I get the message all our circuits are busy now then few second later, the phone rings, going to the second route! And the call can be established, how can I get rid of this message?? thanks ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ACD ASR
Is there a ready add-on to asterisk that will display the ACD/ASR per channel, source destination? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] delay to dial
Hello all, Is there anyway that I can configure Asterisk to start dialing out from fxo after (xx) seconds from getting the dial tone? I don't want tdm card to send the number immediately because it fails many times. Thanks for any help. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] delay to dial
I use elastix,
I have this for dialout:
exten = s,8,Dial(${OUT_${ARG1}}/${ARG2:${length}})
where should I add the w ??
also what If I want 1 second delay?
thanks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Saturday, October 10, 2009 5:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] delay to dial
B.Masoud @ SH wrote:
Hello all,
Is there anyway that I can configure Asterisk to start dialing out
from fxo after (xx) seconds from getting the dial tone? I don't want
tdm card to send the number immediately because it fails many times.
You can use the w. This is from the wiki:
If you need a .5 second pause while dialing a number you can insert a
*w* in the appropriate place.
Example:
exten = _5XXX,n,Dial(ZAP/G1/w1269xxxw${EXTEN}${CALLERID(number)})
This dials out G1, waits 1/2 second, dials the phone number and then
waits 1/2 second again and then dial the extension along with the
callerid number.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] delay to dial
I have done the changes
exten = s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
I am getting this:
-- Executing [...@macro-dialout-trunk:19] Dial(IAX2/9-11592,
DAHDI/r0/0559857826|300|) in new stack
-- Called r0/0559857826
Is it now on work? Or I have to restart?
Thanks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Saturday, October 10, 2009 6:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] delay to dial
Ivan Stepaniuk wrote:
John I think you are wrong, I don't know elastix but the OUT_${ARG1} var
seems to contain the channel technology, the 'w' should be inserted
after the slash.
exten = s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
I agree.
Doug
--
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Safety, deserve neither Liberty nor Safety.
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Re: [asterisk-users] delay to dial
Sorry for keep asking, but I did extensions reload, and restarted asterisk,
What should the message looks like? I still get the same:
-- Executing [...@macro-dialout-trunk:19] Dial(IAX2/9-11592,
DAHDI/r0/0559857826|300|) in new stack
-- Called r0/0559857826
Thanks for your help.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk
Sent: Saturday, October 10, 2009 9:14 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] delay to dial
B.Masoud @ SH wrote:
I have done the changes
exten = s,8,Dial(${OUT_${ARG1}}/www${ARG2:${length}})
I am getting this:
-- Executing [...@macro-dialout-trunk:19] Dial(IAX2/9-11592,
DAHDI/r0/0559857826|300|) in new stack
-- Called r0/0559857826
Is it now on work? Or I have to restart?
It is not working. Issue an 'extensions reload' command at the asterisk
CLI and try again. If it still does not work, then you have edited the
wrong Dial. You should have tried that before asking in the list again.
--
Iván Stepaniuk
Alba Fotónica S.L.
www.albafotonica.com
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[asterisk-users] calls ansowered for 1 second or less
Hello, Sometimes the call gets answered for 1 second, but actually the phone has not rang, its just the CDR, and asterisk hangup automatically, I cought the log of 1 call like this, I hope you can help me with this. My setup is : vendor SIP--à Asterisk ßIAX2---à Asterisk with Dhadi channels Here: -- Executing [966505103...@from-internal:1] Macro(SIP/100-b609f9c0, user-callerid|SKIPTTL|) in new stack -- Executing [...@macro-user-callerid:1] Set(SIP/100-b609f9c0, AMPUSER=100) in new stack -- Executing [...@macro-user-callerid:2] GotoIf(SIP/100-b609f9c0, 0?report) in new stack -- Executing [...@macro-user-callerid:3] ExecIf(SIP/100-b609f9c0, 1|Set|REALCALLERIDNUM=100) in new stack -- Executing [...@macro-user-callerid:4] Set(SIP/100-b609f9c0, AMPUSER=100) in new stack -- Executing [...@macro-user-callerid:5] Set(SIP/100-b609f9c0, AMPUSERCIDNAME=100) in new stack -- Executing [...@macro-user-callerid:6] GotoIf(SIP/100-b609f9c0, 0?report) in new stack -- Executing [...@macro-user-callerid:7] Set(SIP/100-b609f9c0, AMPUSERCID=100) in new stack -- Executing [...@macro-user-callerid:8] Set(SIP/100-b609f9c0, CALLERID(all)=100 100) in new stack -- Executing [...@macro-user-callerid:9] Set(SIP/100-b609f9c0, REALCALLERIDNUM=100) in new stack -- Executing [...@macro-user-callerid:10] ExecIf(SIP/100-b609f9c0, 0|Set|CHANNEL(language)=) in new stack -- Executing [...@macro-user-callerid:11] GotoIf(SIP/100-b609f9c0, 1?continue) in new stack -- Goto (macro-user-callerid,s,20) -- Executing [...@macro-user-callerid:20] NoOp(SIP/100-b609f9c0, Using CallerID 100 100) in new stack -- Executing [966505103...@from-internal:2] Set(SIP/100-b609f9c0, _NODEST=) in new stack -- Executing [966505103...@from-internal:3] Macro(SIP/100-b609f9c0, record-enable|100|OUT|) in new stack -- Executing [...@macro-record-enable:1] GotoIf(SIP/100-b609f9c0, 1?check) in new stack -- Goto (macro-record-enable,s,4) -- Executing [...@macro-record-enable:4] AGI(SIP/100-b609f9c0, recordingcheck|20091009-194302|1255102982.3126) in new stack -- Launched AGI Script /var/lib/asterisk/agi-bin/recordingcheck recordingcheck|20091009-194302|1255102982.3126: Outbound recording not enabled -- AGI Script recordingcheck completed, returning 0 -- Executing [...@macro-record-enable:5] MacroExit(SIP/100-b609f9c0, ) in new stack -- Executing [966505103...@from-internal:4] Macro(SIP/100-b609f9c0, dialout-trunk|12|505103150||) in new stack -- Executing [...@macro-dialout-trunk:1] Set(SIP/100-b609f9c0, DIAL_TRUNK=12) in new stack -- Executing [...@macro-dialout-trunk:2] GosubIf(SIP/100-b609f9c0, 0?sub-pincheck|s|1) in new stack -- Executing [...@macro-dialout-trunk:3] GotoIf(SIP/100-b609f9c0, 0?disabletrunk|1) in new stack -- Executing [...@macro-dialout-trunk:4] Set(SIP/100-b609f9c0, DIAL_NUMBER=505103150) in new stack -- Executing [...@macro-dialout-trunk:5] Set(SIP/100-b609f9c0, DIAL_TRUNK_OPTIONS=trf) in new stack -- Executing [...@macro-dialout-trunk:6] Set(SIP/100-b609f9c0, OUTBOUND_GROUP=OUT_12) in new stack -- Executing [...@macro-dialout-trunk:7] GotoIf(SIP/100-b609f9c0, 1?nomax) in new stack -- Goto (macro-dialout-trunk,s,9) -- Executing [...@macro-dialout-trunk:9] GotoIf(SIP/100-b609f9c0, 0?skipoutcid) in new stack -- Executing [...@macro-dialout-trunk:10] Set(SIP/100-b609f9c0, DIAL_TRUNK_OPTIONS=) in new stack -- Executing [...@macro-dialout-trunk:11] Macro(SIP/100-b609f9c0, outbound-callerid|12) in new stack -- Executing [...@macro-outbound-callerid:1] ExecIf(SIP/100-b609f9c0, 0|SetCallerPres|) in new stack -- Executing [...@macro-outbound-callerid:2] ExecIf(SIP/100-b609f9c0, 0|Set|REALCALLERIDNUM=100) in new stack -- Executing [...@macro-outbound-callerid:3] GotoIf(SIP/100-b609f9c0, 1?normcid) in new stack -- Goto (macro-outbound-callerid,s,6) -- Executing [...@macro-outbound-callerid:6] Set(SIP/100-b609f9c0, USEROUTCID=) in new stack -- Executing [...@macro-outbound-callerid:7] Set(SIP/100-b609f9c0, EMERGENCYCID=) in new stack -- Executing [...@macro-outbound-callerid:8] Set(SIP/100-b609f9c0, TRUNKOUTCID=9) in new stack -- Executing [...@macro-outbound-callerid:9] GotoIf(SIP/100-b609f9c0, 1?trunkcid) in new stack -- Goto (macro-outbound-callerid,s,12) -- Executing [...@macro-outbound-callerid:12] ExecIf(SIP/100-b609f9c0, 1|Set|CALLERID(all)=9) in new stack -- Executing [...@macro-outbound-callerid:13] GotoIf(SIP/100-b609f9c0, 1?exit) in new stack -- Goto (macro-outbound-callerid,s,11) -- Executing [...@macro-outbound-callerid:11] MacroExit(SIP/100-b609f9c0, ) in new stack -- Executing [...@macro-dialout-trunk:12] ExecIf(SIP/100-b609f9c0, 0|AGI|fixlocalprefix) in new stack -- Executing [...@macro-dialout-trunk:13] Set(SIP/100-b609f9c0, OUTNUM=0505103150) in new stack --
[asterisk-users] choppy sound
Hi After a day of running asterisk, I got choppy sound when fw ip-pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] choppy sound
Hi, I am using CentOS Asterisk 1.4 The server has 4GB RAM, 2Ghz Duo Core, and digium 24ports fxo no hardware echo cancelation Does hardware echo will help? Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, October 09, 2009 11:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] choppy sound It would be helpful to know the OS, release of Asterisk, hardware, etc. In my case, I start getting excessive echoes at end of day, so I do a restart when convenient each morning around 4:00 AM. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 09, 2009 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] choppy sound Hi After a day of running asterisk, I got choppy sound when fw ip-pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] choppy sound
By the way, how to schedule auto reboot? thanks From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Friday, October 09, 2009 11:51 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] choppy sound It would be helpful to know the OS, release of Asterisk, hardware, etc. In my case, I start getting excessive echoes at end of day, so I do a restart when convenient each morning around 4:00 AM. _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Friday, October 09, 2009 3:46 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] choppy sound Hi After a day of running asterisk, I got choppy sound when fw ip-pstn When I restart asterisk the sound is fine, Anyone had same problem? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] limiting number of channels to be accessed
Hello all, Assuming I have 1 asterisk with 24 channels fxo and another 2 asterisk boxes all connected iax2, I want to grand the first asterisk box to use all the 24 channels on the main, but I want the 2nd asterisk to use only 8 port, how can limit the second box from receiving more than 8 simultaneous calls?? (even if the main have available ports) Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk debug message --- stopped sounds ???
Anyone pls I have seen this message stopped sounds while I am watching asterisk debug: -- Called 9/0532828384 -- Call accepted by 192.168.10.220 (format ulaw) -- Format for call is ulaw -- IAX2/9-69 stopped sounds -- IAX2/9-69 answered SIP/xxx.xxx.xxx.xxx-b7d009a0 What does it mean?? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] limiting number of channels to be accessed
Where do I add these commands? To which file?
[macro-stdvoip]
; ${ARG1} - full dial string
; Return ${DIALSTATUS} = CHANUNAVAIL if ${VOIPMAX} exceeded
exten = s,1,Set(GROUP()=trunkgroup1) ;Set Group
exten = s,2,GotoIf($[${GROUP_COUNT(trunkgroup1)} ${VOIPMAX}]?103)
;Exceeded?
exten = s,3,Dial(${ARG1}) ;dial it
exten = s,103,SetVar(DIALSTATUS=CHANUNAVAIL) ;deny call
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk
Sent: Thursday, October 08, 2009 12:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] limiting number of channels to be accessed
B.Masoud @ SH wrote:
I want to grand the first asterisk box to use all the 24 channels on the
main, but I want the 2^nd asterisk to use only 8 port, how can limit the
second box from receiving more than 8 simultaneous calls?? (even if the
main have available ports)
This can be done using the GROUP functions under asterisk.
Check this, look at example #2:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+group
--
Iván Stepaniuk
Alba Fotónica S.L.
www.albafotonica.com
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Re: [asterisk-users] limiting number of channels to be accessed
What do you mean by tracking incoming channels???
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Michelle
Dupuis
Sent: Thursday, October 08, 2009 9:11 PM
To: 'Asterisk Users List'
Subject: Re: [asterisk-users] limiting number of channels to be accessed
And how do you track incoming channels on this trunk?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Mathers
Sent: Thursday, October 08, 2009 2:01 PM
To: Asterisk Users List
Subject: Re: [asterisk-users] limiting number of channels to be accessed
/etc/asterisk/extensions.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH
Sent: Thursday, October 08, 2009 11:46 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] limiting number of channels to be accessed
Where do I add these commands? To which file?
[macro-stdvoip]
; ${ARG1} - full dial string
; Return ${DIALSTATUS} = CHANUNAVAIL if ${VOIPMAX} exceeded
exten = s,1,Set(GROUP()=trunkgroup1) ;Set Group
exten = s,2,GotoIf($[${GROUP_COUNT(trunkgroup1)} ${VOIPMAX}]?103)
;Exceeded?
exten = s,3,Dial(${ARG1}) ;dial it
exten = s,103,SetVar(DIALSTATUS=CHANUNAVAIL) ;deny call
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk
Sent: Thursday, October 08, 2009 12:38 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] limiting number of channels to be accessed
B.Masoud @ SH wrote:
I want to grand the first asterisk box to use all the 24 channels on
the main, but I want the 2^nd asterisk to use only 8 port, how can
limit the second box from receiving more than 8 simultaneous calls??
(even if the main have available ports)
This can be done using the GROUP functions under asterisk.
Check this, look at example #2:
http://www.voip-info.org/wiki/index.php?page=Asterisk+func+group
--
Iván Stepaniuk
Alba Fotónica S.L.
www.albafotonica.com
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__ Information from ESET NOD32 Antivirus, version of virus signature
database 4491 (20091008) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
__ Information from ESET NOD32 Antivirus, version of virus signature
database 4491 (20091008) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
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Re: [asterisk-users] tdm outgoing
Thanks, What if I want to group a TDM2400 into 3 groups, r0/0 to r0/7 , r1/8 to r1/15 , r2/16 to r2/23 How to do that? Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Philipp Kempgen Sent: Monday, October 05, 2009 10:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm outgoing B.Masoud @ SH schrieb: I have defined the card g0 to have 24 channels, but every time I try to call, if all ports are off the call always go to the first port, how can I balance the calls over all ports??? http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels#DialingaGroup Dial(Dahdi/r0/...) Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk debug message --- stopped sounds ???
I have seen this message stopped sounds while I am watching asterisk debug: -- Called 9/0532828384 -- Call accepted by 192.168.10.220 (format ulaw) -- Format for call is ulaw -- IAX2/9-69 stopped sounds -- IAX2/9-69 answered SIP/xxx.xxx.xxx.xxx-b7d009a0 What does it mean?? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Networking Concept
China too wide, but regardless! How is asterisk take care such situation? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk Sent: Tuesday, October 06, 2009 2:36 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Networking Concept B.Masoud @ SH wrote: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk will deal with this call?? Will his latency be JAPN-UK + UK-China (around 1000ms !) or only from Japan to China??? In the case of the SIP protocol, the audio (RTP) traffic can be re-routed on the fly from A(jp) to C(ch), reducing the audio latency, (and sometimes increasing your headaches). This is calling re-INVITE, and can be turned on on asterisk. For other protocols there are similar features. I think your latency figures are a little bit exaggerated if you speak about the network latency. I am in Spain and my latency to China at my home ADSL is arround 80ms for mainland. 250ms to Tokio tough. Regards -- Iván Stepaniuk Alba Fotónica S.L. www.albafotonica.com ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Networking Concept
How they can? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Neland Sent: Tuesday, October 06, 2009 12:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Networking Concept - Original Message - From: B.Masoud @ SH mailto:i...@saudihome.com To: 'Asterisk Users Mailing List - Non-Commercial mailto:asterisk-users@lists.digium.com Discussion' Sent: Tuesday, October 06, 2009 1:14 AM Subject: [asterisk-users] Networking Concept Hello, I would like to know how Asterisk deal in this case: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk will deal with this call?? Will his latency be JAPN-UK + UK-China (around 1000ms !) or only from Japan to China??? Be sure not to run into trouble for running inlicenced ip-telephony in China, so the government can't (as easily) intercept your calls. Leif ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
/100-08fba098, 0|Set|DIAL_TRUNK_OPTIONS=M(setmusic^)) in new stack -- Executing [...@macro-dialout-trunk:16] Macro(SIP/100-08fba098, dialout-trunk-predial-hook|) in new stack -- Executing [...@macro-dialout-trunk-predial-hook:1] MacroExit(SIP/100-08fba098, ) in new stack -- Executing [...@macro-dialout-trunk:17] GotoIf(SIP/100-08fba098, 0?bypass|1) in new stack -- Executing [...@macro-dialout-trunk:18] GotoIf(SIP/100-08fba098, 0?customtrunk) in new stack -- Executing [...@macro-dialout-trunk:19] Dial(SIP/100-08fba098, DAHDI/DGTDM24/966505103250|300|) in new stack == Everyone is busy/congested at this time (1:0/0/1) -- Executing [...@macro-dialout-trunk:20] Goto(SIP/100-08fba098, s-CHANUNAVAIL|1) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,1) -- Executing [s-chanunav...@macro-dialout-trunk:1] GotoIf(SIP/100-08fba098, 1?noreport) in new stack -- Goto (macro-dialout-trunk,s-CHANUNAVAIL,3) -- Executing [s-chanunav...@macro-dialout-trunk:3] NoOp(SIP/100-08fba098, TRUNK Dial failed due to CHANUNAVAIL (hangupcause: 0) - failing through to other trunks) in new stack -- Executing [966505103...@from-internal:5] Macro(SIP/100-08fba098, outisbusy|) in new stack -- Executing [...@macro-outisbusy:1] Playback(SIP/100-08fba098, all-circuits-busy-now|noanswer) in new stack -- SIP/100-08fba098 Playing 'all-circuits-busy-now' (language 'en') -- Executing [...@macro-outisbusy:2] Playback(SIP/100-08fba098, pls-try-call-later|noanswer) in new stack -- SIP/100-08fba098 Playing 'pls-try-call-later' (language 'en') == Spawn extension (macro-outisbusy, s, 2) exited non-zero on 'SIP/100-08fba098' in macro 'outisbusy' == Spawn extension (from-internal, 966505103250, 5) exited non-zero on 'SIP/100-08fba098' -- Executing [...@from-internal:1] Macro(SIP/100-08fba098, hangupcall) in new stack -- Executing [...@macro-hangupcall:1] ResetCDR(SIP/100-08fba098, w) in new stack -- Executing [...@macro-hangupcall:2] NoCDR(SIP/100-08fba098, ) in new stack -- Executing [...@macro-hangupcall:3] GotoIf(SIP/100-08fba098, 1?skiprg) in new stack -- Goto (macro-hangupcall,s,6) -- Executing [...@macro-hangupcall:6] GotoIf(SIP/100-08fba098, 1?skipblkvm) in new stack -- Goto (macro-hangupcall,s,9) -- Executing [...@macro-hangupcall:9] GotoIf(SIP/100-08fba098, 1?theend) in new stack -- Goto (macro-hangupcall,s,11) -- Executing [...@macro-hangupcall:11] Hangup(SIP/100-08fba098, ) in new stack == Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/100-08fba098' in macro 'hangupcall' == Spawn extension (from-internal, h, 1) exited non-zero on 'SIP/100-08fba098' elastix*CLI please let me know what is wrong??? Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Paul Hales Sent: Monday, October 05, 2009 2:38 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm outgoing Is inbound working? Can you see action on the CLI when you send a call to the lines attached to the card? PaulH B.Masoud @ SH wrote: Hi I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls to that trunk, I am getting all circuits are busy now, do I have to do something specific?? I am using elastix. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
Are you series??? My card is FXO TDM2400, I am sure its designed to forward calls to pstn!!! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ira Sent: Monday, October 05, 2009 5:07 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm outgoing At 04:32 PM 10/4/2009, you wrote: Hi I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls to that trunk, I am getting all circuits are busy now, do I have to do something specific?? I am using elastix. Sometimes you can't make a call on DAHDI until a call has been received. At least I can't. Ira ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
Man, thanks a lot! I just changed the name to g0 instead of DGTDM24 and it worked!! I would like to know where I can set the configuration for line tones( dial tone, call and busy tone) and where I can change different setting for polarity / current disconnect etc.. of the line? I cant find Zapata.cfg Thanks again! -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ioan Indreias Sent: Monday, October 05, 2009 1:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm outgoing DAHDI/DGTDM24/966505103250 This (DGTDM24) is strange. Could you provide the setup of the DAHDI trunk? You should have something like DAHDI/g0/96 or DAHDI/10/96 Here are more info on this subject: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg226642.html HTH, Ioan (Nini) Indreias www.modulo.ro ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] tdm outgoing
Thanks, I made the zone, and now call disconnect works ok! i have one last problem, I have defined the card g0 to have 24 channels, but every time I try to call, if all ports are off the call always go to the first port, how can I balance the calls over all ports??? Any suggestions appreciated. Thanks all for the help. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ioan Indreias Sent: Monday, October 05, 2009 5:43 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] tdm outgoing I cant find Zapata.cfg You have a DAHDI installation thus you have to find chan_dahdi.conf. it should be located under /etc/asterisk Regarding the configuration for tones you have to check indications.conf file Best regards, Nini ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Networking Concept
Hello, I would like to know how Asterisk deal in this case: Assume I have a Main Asterisk Server located in UK, and another box that have PSTN interfaces located in China, now the purpose is to FW calls through PSTN. Assuming I have a client who is calling from Japan to my main switch in UK and he is calling China, (japan have latency around 500ms to UK and 100ms to China), how asterisk will deal with this call?? Will his latency be JAPN-UK + UK-China (around 1000ms !) or only from Japan to China??? Please let me know. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] tdm outgoing
Hi I installed TDM24 card, made ZAP (DAHDI) trunk, and set outbound all calls to that trunk, I am getting all circuits are busy now, do I have to do something specific?? I am using elastix. Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Trunk Sequence
I have added 2 trunk sequence in my outbound routes, The problem is that: 1. If the call was busy on the first trunk it will go to the second (i.e. the called party hung-up without answering the call) How to overcome this??? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RESET CDR
I use GranStream FXO.. Do you suggest a gateway? Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk Sent: Thursday, September 10, 2009 6:38 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RESET CDR B.Masoud @ SH wrote: Yes that is the problem. So what do you do when the line doesn't support polarity?? What is the best solution in this case? What kind of gateway do you use to connect to the PSTN? -- Iván Stepaniuk Alba Fotónica S.L. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ASR ACD
Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ASR ACD
ASR: Average Success Rate ACD: Average Call duration Thanks. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, September 10, 2009 11:24 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] ASR ACD Please elaborate on what ASR and ACD are. I assume they are not the googled values of Automatic Speech Recognition and automatic call detection. You might want to check out indiosoft.com _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of B.Masoud @ SH Sent: Thursday, September 10, 2009 3:16 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] ASR ACD Is there any program Asterisk users use to calculate ASR and ACD ?? Thanks for any comments. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RESET CDR
I don't want to bill the first 30 seconds, that's all, why is that so strange??? My line does not support polarity reversal, so I don't want to bill for ringing the line... Do you suggest different way than this? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Wednesday, September 09, 2009 8:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RESET CDR On 9/09/09 5:14 PM, B.Masoud @ SH wrote: A little more help is appreciated, I know about ResetCDR() , but I want some code that resets the call data after 30 seconds! And where to put the code exactly. What a strange request. Why exactly are you wanting to do this? If you're wanting all your calls to look like they are 30 seconds shorter can't you just use the time-30 seconds? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RESET CDR
Can you provide me some code for that? I am NOOB -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of David Backeberg Sent: Wednesday, September 09, 2009 5:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RESET CDR On Wed, Sep 9, 2009 at 10:12 AM, B.Masoud @ SHi...@saudihome.com wrote: I don't want to bill the first 30 seconds, that's all, why is that so strange??? My line does not support polarity reversal, so I don't want to bill for ringing the line... Do you suggest different way than this? yes. Subtract 30 seconds from the billing when the call is completed. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RESET CDR
Yes that is the problem. So what do you do when the line doesn't support polarity?? What is the best solution in this case? Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ivan Stepaniuk Sent: Thursday, September 10, 2009 12:02 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RESET CDR Todd Routhier wrote: billsecs is a field in the CDR, it's already there.. Just don't bill based on the duration field, bill based on the billsecs field and you should have what you want. He says that the line does not support polarity reversal. It really depends on the type of PSTN interface he is using, but this probably means that the duration and the billsecs fields are going to be the same, as the channel gets answered and the ring-back tone is also counted. I would rather say that if you do not have a proper way to detect the call progress, the system is not reliable for billing your users. The main problem with the 30 seconds solution is that you will have a lot of calls made without charge (ie: 3 seconds ringing, 20 seconds call). And you will also charge people for ringing a phone during more that 30 seconds (ie: calling my grandma) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] RESET CDR
Hello, How can I reset CDR time , let's say after 30 seconds of answer signal, reset CDR to 0 , any idea ?? Thanks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RESET CDR
A little more help is appreciated, I know about ResetCDR() , but I want some code that resets the call data after 30 seconds! And where to put the code exactly. Thanks. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Matt Riddell Sent: Wednesday, September 09, 2009 7:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RESET CDR On 9/09/09 4:34 PM, B.Masoud @ SH wrote: Hello, How can I reset CDR time , let's say after 30 seconds of answer signal, reset CDR to 0 , any idea ?? :) Use the ResetCDR application? -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
