[asterisk-users] Problems Extension with a Call In on Asterisk 1.6
Hi I request your help because i don't have actually a solution at my problems. I have a Asterisk Server in 1.6 Connected at a SIP Provider This provider supply me 2 numbers: 003318364 (official number) 081169 (Nddi Number) When i receive a call on the 081169, he don't use the extension. He use the 003318364 extension. SIP Debug: --- SIP read from UDP://91.121.xxx.xxx:5060 --- INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 Allow: UPDATE,REFER,INFO Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Contact: sip:91.121.xxx.xxx:5060 Content-Type: application/sdp CSeq: 1602837515 INVITE From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 P-Preferred-Identity: sip:033426aaa...@sip.myoperator.net;user=phone To: sip:081169x...@91.121.xxx.xxx;user=phone User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 Content-Length: 481 v=0 o=cp10 130085910854 130085910854 IN IP4 10.7.1.121 s=SIP Call c=IN IP4 91.121.bbb.bbb t=0 0 m=audio 36146 RTP/AVP 18 4 0 8 125 111 101 b=AS:21 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:125 CLEARMODE/8000/1 a=rtpmap:111 iLBC/8000/1 a=fmtp:111 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=sqn:0 a=cdsc: 1 image udptl t38 - --- (13 headers 22 lines) --- Sending to 91.121.xxx.xxx : 5060 (no NAT) Using INVITE request as basis request - 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Found peer 'Myoperator' for '033426aa' from 91.121.xxx.xxx:5060 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 125 Found RTP audio format 111 Found RTP audio format 101 Peer audio RTP is at port 91.121.bbb.bbb:36146 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found unknown media description format CLEARMODE for ID 125 Found audio description format iLBC for ID 111 Found audio description format telephone-event for ID 101 Capabilities: us - 0x109 (g723|alaw|g729), peer - audio=0x50d (g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x109 (g723|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 91.121.bbb.bbb:36146 Looking for 003318364 in Appels-Entrants (domain 78.41.xxx.xxx) --- Reliably Transmitting (no NAT) to 91.121.xxx.xxx:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2;received=91.121.xxx.xxx From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net CSeq: 1602837515 INVITE Server: Asterisk PBX 1.6.1.8 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527 handle_request_invite: Call from '0033459aa' to extension '003318364' rejected because extension not found. Scheduling destruction of SIP dialog '04459-nk-5fa6f8a0-18641f...@sip.myoperator.net' in 6400 ms (Method: INVITE) --- SIP read from UDP://91.121.xxx.xxx:5060 --- ACK sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Contact: sip:91.121.xxx.xxx:5060 CSeq: 1602837515 ACK From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 Content-Length: 0 I see in the debug: To: sip:081169x...@91.121.xxx.xxx;user=phone but he search the 003318364 extension [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527 handle_request_invite: Call from '0033459aa' to extension '003318364' rejected because extension not found. Anyone know the solution for he use the extension based on the To: ? thanks Olivier -- _ -- 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] Problems Extension with a Call In on Asterisk 1.6
Hi Oliver , This is a simple scenario with asterisk you can edit sip.conf and in peer entry, try to add, context=(desired_context for peer) and then into context write a dial-plan for given number and route a call or whatever you want to do. On Wed, Mar 23, 2011 at 11:31 AM, Olivier CALVANO o.calv...@gmail.comwrote: Hi I request your help because i don't have actually a solution at my problems. I have a Asterisk Server in 1.6 Connected at a SIP Provider This provider supply me 2 numbers: 003318364 (official number) 081169 (Nddi Number) When i receive a call on the 081169, he don't use the extension. He use the 003318364 extension. SIP Debug: --- SIP read from UDP://91.121.xxx.xxx:5060 --- INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 Allow: UPDATE,REFER,INFO Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Contact: sip:91.121.xxx.xxx:5060 Content-Type: application/sdp CSeq: 1602837515 INVITE From: 033426aa sip:033426aaa...@sip.myoperator.net ;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 P-Preferred-Identity: sip:033426aaa...@sip.myoperator.net;user=phone To: sip:081169x...@91.121.xxx.xxx;user=phone User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 Content-Length: 481 v=0 o=cp10 130085910854 130085910854 IN IP4 10.7.1.121 s=SIP Call c=IN IP4 91.121.bbb.bbb t=0 0 m=audio 36146 RTP/AVP 18 4 0 8 125 111 101 b=AS:21 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:125 CLEARMODE/8000/1 a=rtpmap:111 iLBC/8000/1 a=fmtp:111 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=sqn:0 a=cdsc: 1 image udptl t38 - --- (13 headers 22 lines) --- Sending to 91.121.xxx.xxx : 5060 (no NAT) Using INVITE request as basis request - 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Found peer 'Myoperator' for '033426aa' from 91.121.xxx.xxx:5060 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 125 Found RTP audio format 111 Found RTP audio format 101 Peer audio RTP is at port 91.121.bbb.bbb:36146 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found unknown media description format CLEARMODE for ID 125 Found audio description format iLBC for ID 111 Found audio description format telephone-event for ID 101 Capabilities: us - 0x109 (g723|alaw|g729), peer - audio=0x50d (g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x109 (g723|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 91.121.bbb.bbb:36146 Looking for 003318364 in Appels-Entrants (domain 78.41.xxx.xxx) --- Reliably Transmitting (no NAT) to 91.121.xxx.xxx:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2;received=91.121.xxx.xxx From: 033426aa sip:033426aaa...@sip.myoperator.net ;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net CSeq: 1602837515 INVITE Server: Asterisk PBX 1.6.1.8 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527 handle_request_invite: Call from '0033459aa' to extension '003318364' rejected because extension not found. Scheduling destruction of SIP dialog '04459-nk-5fa6f8a0-18641f...@sip.myoperator.net' in 6400 ms (Method: INVITE) --- SIP read from UDP://91.121.xxx.xxx:5060 --- ACK sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Contact: sip:91.121.xxx.xxx:5060 CSeq: 1602837515 ACK From: 033426aa sip:033426aaa...@sip.myoperator.net ;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 Content-Length: 0 I see in the debug: To: sip:081169x...@91.121.xxx.xxx;user=phone but he search the 003318364 extension [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527 handle_request_invite: Call from '0033459aa' to extension '003318364' rejected because extension not found. Anyone know the solution for he use the extension based on the To: ? thanks Olivier -- _ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] Sangoma wapipe installation error
Try: cd /usr/src/dahdi ./Setup dahdi That's it. Am 22.03.2011 21:06, schrieb satish patel: Hey! I am installing Sangoma A102D wanpipe driver and i got following error. what is this ? why dir isn't there ? wanpipe-3.5.16 # make dahdi DAHDI_DIR=/usr/src/dahdi wanpipe-3.5.16 # make install Send Installing Wanpipe Firmware update utility in /etc/wanpipe/util/wan_aftup install -D wan_aftup /usr/sbin/wan_aftup install -d /etc/wanpipe/util/wan_aftup/scripts install: cannot create directory `/etc/wanpipe/util/wan_aftup': Not a directory make[2]: *** [install] Error 1 make[2]: Leaving directory `/usr/local/src/asterisk/wanpipe-3.5.16/util/wan_aftup' make[1]: *** [install] Error 2 make[1]: Leaving directory `/usr/local/src/asterisk/wanpipe-3.5.16/util' make: *** [install_util] Error 2 -- _ -- 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] Problems Extension with a Call In on Asterisk 1.6
Hi Dhaval, Thanks for your answer, but i not my question ;=) My asterisk have a entry into the sip.conf with a context. in extensions.conf, i have this extensions: exten = _003318364,1,Dial(SIP/203,180,rt) exten = _003381169,1,Dial(SIP/204,180,rt) (in my debug, i have deleted the exten = _003318364) When i call to 3318364 that's work When i call to 3381169 that's work but it's the _003318364 is used and phone 203 ring bye olivier 2011/3/23 DHAVAL INDRODIYA dhaval.it01...@gmail.com: Hi Oliver , This is a simple scenario with asterisk you can edit sip.conf and in peer entry, try to add, context=(desired_context for peer) and then into context write a dial-plan for given number and route a call or whatever you want to do. On Wed, Mar 23, 2011 at 11:31 AM, Olivier CALVANO o.calv...@gmail.com wrote: Hi I request your help because i don't have actually a solution at my problems. I have a Asterisk Server in 1.6 Connected at a SIP Provider This provider supply me 2 numbers: 003318364 (official number) 081169 (Nddi Number) When i receive a call on the 081169, he don't use the extension. He use the 003318364 extension. SIP Debug: --- SIP read from UDP://91.121.xxx.xxx:5060 --- INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 Allow: UPDATE,REFER,INFO Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Contact: sip:91.121.xxx.xxx:5060 Content-Type: application/sdp CSeq: 1602837515 INVITE From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 P-Preferred-Identity: sip:033426aaa...@sip.myoperator.net;user=phone To: sip:081169x...@91.121.xxx.xxx;user=phone User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 Content-Length: 481 v=0 o=cp10 130085910854 130085910854 IN IP4 10.7.1.121 s=SIP Call c=IN IP4 91.121.bbb.bbb t=0 0 m=audio 36146 RTP/AVP 18 4 0 8 125 111 101 b=AS:21 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:125 CLEARMODE/8000/1 a=rtpmap:111 iLBC/8000/1 a=fmtp:111 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=sqn:0 a=cdsc: 1 image udptl t38 - --- (13 headers 22 lines) --- Sending to 91.121.xxx.xxx : 5060 (no NAT) Using INVITE request as basis request - 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Found peer 'Myoperator' for '033426aa' from 91.121.xxx.xxx:5060 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 125 Found RTP audio format 111 Found RTP audio format 101 Peer audio RTP is at port 91.121.bbb.bbb:36146 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found unknown media description format CLEARMODE for ID 125 Found audio description format iLBC for ID 111 Found audio description format telephone-event for ID 101 Capabilities: us - 0x109 (g723|alaw|g729), peer - audio=0x50d (g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x109 (g723|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 91.121.bbb.bbb:36146 Looking for 003318364 in Appels-Entrants (domain 78.41.xxx.xxx) --- Reliably Transmitting (no NAT) to 91.121.xxx.xxx:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2;received=91.121.xxx.xxx From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net CSeq: 1602837515 INVITE Server: Asterisk PBX 1.6.1.8 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527 handle_request_invite: Call from '0033459aa' to extension '003318364' rejected because extension not found. Scheduling destruction of SIP dialog '04459-nk-5fa6f8a0-18641f...@sip.myoperator.net' in 6400 ms (Method: INVITE) --- SIP read from UDP://91.121.xxx.xxx:5060 --- ACK sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Contact: sip:91.121.xxx.xxx:5060 CSeq: 1602837515 ACK From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP
Re: [asterisk-users] wrong time retrieved from system command
Tilghman,
Could you remove your Reply-By header, please? Your deadline is two months in
the past (and in any case, list postings really shouldn't have a reply deadline
at all)
Here is your Reply-By header from your March 21 email:
Reply-By: Wed, 19 Jan 2011 16:20:00 -0600
Thanks!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman Lesher
Sent: Monday, March 21, 2011 3:54 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] wrong time retrieved from system command
On Monday 21 March 2011 06:45:37 asterisk asterisk wrote:
${STRFTIME(${EPOCH},GMT+8,%G%m%d-%H%M%S)}
I use the above command to get the system date and time
it returns 20110321-034329
but it is exactly 8 hours early than the system time when I type date
in linux terminal
Mon Mar 21 19:43:35 HKT 2011
I am looking for help.
Do you have an file (or symlink) in /usr/share/zoneinfo called GMT+8? I
certainly don't, and I'm not running anything different from the standard set
of zone files. If you don't have that entry, then the timezone code will use
UTC (i.e. no local differentiations).
--
Tilghman
--
_
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New to Asterisk? Join us for a live introductory webinar every Thurs:
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asterisk-users mailing list
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[asterisk-users] Asterisk using as a SIP client
Hi all I am planning to use asterisk as a IP phone(Porting asterisk into a hardware).Is there any limitations if I use asterisk as a SIP client?,and asterisk has any advantages if use like this? Thanks Nikhil -- _ -- 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 using as a SIP client
On 23 Mar 2011, at 10:40, Nikhil wrote: I am planning to use asterisk as a IP phone(Porting asterisk into a hardware). Interesting.. Is there any limitations if I use asterisk as a SIP client?,and asterisk has any advantages if use like this? It's not really designed as a SIP client. It's probably possible, but it's very sledgehammer. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk using as a SIP client
i agree. first time im hearing this. not saying its not possible but the things you will need to do to make it work it wouldnt be worthwhile.. Asterisk PBX is more of a server relaying SIP/SCCP/IAX/DAHDI calls and having softphone or hardware phone clients connecting to it. John. On Wed, 2011-03-23 at 11:18 +, Steven Howes wrote: On 23 Mar 2011, at 10:40, Nikhil wrote: I am planning to use asterisk as a IP phone(Porting asterisk into a hardware). Interesting.. Is there any limitations if I use asterisk as a SIP client?,and asterisk has any advantages if use like this? It's not really designed as a SIP client. It's probably possible, but it's very sledgehammer. S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- 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] Sangoma wapipe installation error
Hey, I did ./Setup dahdi and everything went well but i didn't find any command wancfg_dahdi -- WANPIPE v3.5.19 Installation Script Copyright (c) 1995-2010, Sangoma Technologies Inc. -- WANPIPE META CONFIGURATION There are two configuration files associated with WANPIPE. 1) /usr/local/src/asterisk/wanpipe-3.5.19/wanrouter.rc: - defines locations of important files such as lock and configuration files as well as start/stop order of multiple WANPIPE devices. 2) /usr/local/src/asterisk/wanpipe-3.5.19/wanpipe1.conf: - main configuration file for each WANPIPE device. - defines interfaces, hardware and protocol information. - this file can be created using the 'wancfg' GUI utility or manually based on sample files located in /etc/wanpipe/samples. Please read the WanpipeInstallation.(pdf/txt) manual for further information. Wanpipe META config file found in /etc/wanpipe directory Wanpipe startup sequence: wanpipe1 -- WANPIPE v3.5.19 Installation Script Copyright (c) 1995-2010, Sangoma Technologies Inc. -- WANPIPE UTILITIES SETUP WANPIPE utilities are used to: 1) create configuration files: for Zaptel and Asterisk /usr/sbin/wancfg_zaptel #Zaptel and Asterisk /usr/sbin/wancfg_dahdi #Dahdi and Asterisk /usr/sbin/wancfg_smg#BRI/SS7, Zaptel and Asterisk /usr/sbin/wancfg_tdmapi #TDM API 2) create WANPIPE WAN/IP configuration files. (/usr/sbin/wancfg) 3) start,stop,restart individual/all devices and interfaces. (/usr/sbin/wanrouter) 4) debug line, protocol and driver problems. (/usr/sbin/wanpipemon) 5) aid in WANPIPE API development (/etc/wanpipe/api) Refer to the WanpipeInstallation.(pdf/txt) for more information. Compiling WANPIPE Utilities ... Done. Compiling WANPIPE WanCfg Utility ...Done. Compiling WANPIPE LibSangoma API library ...Done. Compiling WANPIPE LibStelephony API library ...Done. Compiling WANPIPE API Development Utilities ...Done. Compiling WANPIPE HWEC Utilities ...Done. WANPIPE Environment Setup Complete !!! Installing WANPIPE Files ... ! Installing WANPIPE Utilities in /usr/sbin Installing wanrouter.rc in /etc/wanpipe Installing wanpipe libraries in /etc/wanpipe Installing firmware in /etc/wanpipe/firmware Installing documentation in /usr/share/doc/wanpipe Installing sample api code in /etc/wanpipe/api Installing AFT Firmware update utility in /etc/wanpipe/util cp: cannot overwrite non-directory `/etc/wanpipe/util/wan_aftup' with directory `util/wan_aftup/' Installing driver headers in /etc/wanpipe/api/include/linux Installing Hardware Echo Cancel Utilites -- WANPIPE v3.5.19 Installation Script Copyright (c) 1995-2010, Sangoma Technologies Inc. -- WANPIPE INSTALLATON: COMPLETE WANPIPE installation is now complete. WANPIPE kernel drivers and configuration/debug utilities have been compiled and installed. 1) Proceed to configure the WANPIPE drivers: Asterisk/Zaptel : /usr/sbin/wancfg_zaptel Asterisk/Dahdi : /usr/sbin/wancfg_dahdi TDM API : /usr/sbin/wancfg_tdmapi SMG SS7/BRI/PRI : /usr/sbin/wancfg_smg WAN Routing/API : /usr/sbin/wancfg 2) Use the /usr/sbin/wanrouter startup script to start and stop the router. (eg: wanrouter start) 3) To uninstall WANPIPE package run ./Setup remove Please read http://wiki.sangoma.com for further instructions. root@example:/usr/local/src/asterisk/wanpipe-3.5.19# wancfg_dahdi wancfg_dahdi: command not found root@example:/usr/local/src/asterisk/wanpipe-3.5.19# Date: Wed, 23 Mar 2011 09:28:25 +0100 From: t...@ovm-group.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sangoma wapipe installation error Try: cd /usr/src/dahdi ./Setup dahdi That's it. Am 22.03.2011 21:06, schrieb satish patel: Hey! I am installing Sangoma A102D wanpipe driver and i got following error. what is this ? why dir isn't there ? wanpipe-3.5.16 # make dahdi DAHDI_DIR=/usr/src/dahdi wanpipe-3.5.16 # make install Send Installing Wanpipe Firmware update utility in /etc/wanpipe/util/wan_aftup install -D wan_aftup
Re: [asterisk-users] IAX Call token revisited
Kevin wrote: On 03/21/2011 06:49 PM, Dan Austin wrote: I just finished a fresh install of 1.8.3.2 at home using the packages Digium hosts. After correcting a number of typo/config'o error that had crept in over the years, I thought I had everything working. My wife just complained that she cannot call her mother (who is using an old IAX hardphone I left for her). After turning up the logging level I see- chan_iax2.c: Call rejected, CallToken Support required Which google cays can be fixed with: [general] calltokenoptional=0.0.0.0/0.0.0.0 maxcallnumbers=16384 or [peer] requirecalltoken=no (or auto) Either set of changes does suppress the error, but the remote device still fails to register. No other errors/warnings are present. If there aren't any errors or warnings appearing, then you must not have the logging verbosity set high enough. Ensure that you've used 'core set verbose 10' and 'core set debug 10', and that your 'console' channel in logger.conf has all the logger levels enabled. If you still don't see what you are looking for, use 'iax2 set debug' to enable IAX2-specific debugging for that phone's IP address. I should have said relevant errors/warnings. I see info about devastate and queues, but little else. That said I think the problem is unrelated to call token and an issue with the NAT firewall at my mother-in-laws. The incoming traffic is on a very high port and not 4569. I interpret the following log as her phone is not receiving the replies- Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 14807 DCall: 0 [120.xxx.xxx.12:22686] USERNAME: XXX REFRESH : 60 Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00013ms SCall: 03287 DCall: 14807 [120.xxx.xxx.12:22686] AUTHMETHODS : 3 CHALLENGE : \x34\x36\x37\x38\x33\x35\x33\x33 USERNAME: XXX Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 14807 DCall: 0 [120.xxx.xxx.12:22686] USERNAME: XXX REFRESH : 60 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 03287 DCall: 14807 [120.xxx.xxx.12:22686] Tx-Frame Retry[001] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: REGAUTH Timestamp: 00013ms SCall: 03287 DCall: 14807 [120.xxx.xxx.12:22686] AUTHMETHODS : 3 CHALLENGE : \x34\x36\x37\x38\x33\x35\x33\x33 USERNAME: XXX Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 14807 DCall: 0 [120.xxx.xxx.12:22686] USERNAME: XXX REFRESH : 60 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 03287 DCall: 14807 [120.xxx.xxx.12:22686] Rx-Frame Retry[Yes] -- OSeqno: 000 ISeqno: 000 Type: IAX Subclass: REGREQ Timestamp: 3ms SCall: 14807 DCall: 0 [120.xxx.xxx.12:22686] USERNAME: XXX REFRESH : 60 Tx-Frame Retry[-01] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: ACK Timestamp: 3ms SCall: 03287 DCall: 14807 [120.xxx.xxx.12:22686] Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: LAGRQ Timestamp: 10015ms SCall: 03287 DCall: 14807 [120.xxx.xxx.12:22686] I've asked my wife to have her mother reboot her router and phone, but that has not happened yet. Dan -- _ -- 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] What is the most stable version of asterisk?
1.2? 1.4? 1.6? 1.8? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 -- _ -- 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] What is the most stable version of asterisk?
-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Wednesday, March 23, 2011 11:59 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] What is the most stable version of asterisk? 1.2? 1.4? 1.6? 1.8? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 I would vote for 1.4, but keep in mind that it is approaching end-of-life (although that may or may not be relevant, since there are still plenty of 1.2 folks out there). -- _ -- 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] What is the most stable version of asterisk?
I feel there all pretty stable, 1.4,1.6,1.8 depending on how deep of a feature set you want.. if you just want the bare essentials, go with 1.2, but if you want faxing/gTalk, then go with 1.8 I have been running 1.4,1.6,1.8 in production environments and have not have had any serious issues. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, March 23, 2011 1:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] What is the most stable version of asterisk? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Douglas Mortensen Sent: Wednesday, March 23, 2011 11:59 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] What is the most stable version of asterisk? 1.2? 1.4? 1.6? 1.8? Thanks, - Doug Mortensen Network Consultant Impala Networks Inc CCNA, MCSA, Security+, A+ Linux+, Network+, Server+ . www.impalanetworks.com P: (505) 327-7300 F: (505) 327-7545 I would vote for 1.4, but keep in mind that it is approaching end-of- life (although that may or may not be relevant, since there are still plenty of 1.2 folks out there). -- _ -- 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
[asterisk-users] Sangoma A102D wanpiple issue with dahdi
Hey Guy, I have ubuntu 10.04 64bit and compiled dahdi / wanpipe 3.5.19 /asterisk-1.8.x I didn't understand what is the relation between wanpipe and dahdi ? do i need to start wanrouter service ? I am getting weird errors and my system got kernel panic many time when i restart dahdi service. any idea ? what is the startup sequence of all these service ? root@example:/etc/asterisk# /etc/init.d/dahdi stop Unloading DAHDI hardware modules: ERROR: Module wanpipe is in use ERROR: Module dahdi_echocan_mg2 is in use ERROR: Module dahdi is in use by dahdi_echocan_mg2,wanpipe error root@example:/etc/asterisk# wanrouter stop Shutting down wanpipe1 interface: w1g1 Shutting down device: wanpipe2 Shutting down device: wanpipe1 wanconfig: WAN device wanpipe1 did not shutdown : ioctl(wanpipe1,ROUTER_DOWN) failed: : 16 - Device or resource busy If you router was not running ignore this message !! Otherwise, check the /var/log/wanrouter and /var/log/messages for errors Devices Still Running: wanpipe1 -- _ -- 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] Sangoma A102D wanpiple issue with dahdi
added: what is this error ? root@shirley:~# /etc/init.d/dahdi restart Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: wanpipe: error No hardware timing source found in /proc/dahdi, loading dahdi_dummy Running dahdi_cfg: . From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Wed, 23 Mar 2011 17:56:42 + Subject: [asterisk-users] Sangoma A102D wanpiple issue with dahdi Hey Guy, I have ubuntu 10.04 64bit and compiled dahdi / wanpipe 3.5.19 /asterisk-1.8.x I didn't understand what is the relation between wanpipe and dahdi ? do i need to start wanrouter service ? I am getting weird errors and my system got kernel panic many time when i restart dahdi service. any idea ? what is the startup sequence of all these service ? root@example:/etc/asterisk# /etc/init.d/dahdi stop Unloading DAHDI hardware modules: ERROR: Module wanpipe is in use ERROR: Module dahdi_echocan_mg2 is in use ERROR: Module dahdi is in use by dahdi_echocan_mg2,wanpipe error root@example:/etc/asterisk# wanrouter stop Shutting down wanpipe1 interface: w1g1 Shutting down device: wanpipe2 Shutting down device: wanpipe1 wanconfig: WAN device wanpipe1 did not shutdown : ioctl(wanpipe1,ROUTER_DOWN) failed: : 16 - Device or resource busy If you router was not running ignore this message !! Otherwise, check the /var/log/wanrouter and /var/log/messages for errors Devices Still Running: wanpipe1 -- _ -- 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] IAX Call token revisited
On 03/23/2011 11:24 AM, Dan Austin wrote: Kevin wrote: On 03/21/2011 06:49 PM, Dan Austin wrote: I just finished a fresh install of 1.8.3.2 at home using the packages Digium hosts. After correcting a number of typo/config'o error that had crept in over the years, I thought I had everything working. My wife just complained that she cannot call her mother (who is using an old IAX hardphone I left for her). After turning up the logging level I see- chan_iax2.c: Call rejected, CallToken Support required Which google cays can be fixed with: [general] calltokenoptional=0.0.0.0/0.0.0.0 maxcallnumbers=16384 or [peer] requirecalltoken=no (or auto) Either set of changes does suppress the error, but the remote device still fails to register. No other errors/warnings are present. If there aren't any errors or warnings appearing, then you must not have the logging verbosity set high enough. Ensure that you've used 'core set verbose 10' and 'core set debug 10', and that your 'console' channel in logger.conf has all the logger levels enabled. If you still don't see what you are looking for, use 'iax2 set debug' to enable IAX2-specific debugging for that phone's IP address. I should have said relevant errors/warnings. I see info about devastate and queues, but little else. That said I think the problem is unrelated to call token and an issue with the NAT firewall at my mother-in-laws. The incoming traffic is on a very high port and not 4569. I interpret the following log as her phone is not receiving the replies- It's pretty common for an IAX2 device behind a NAT to not be using port 4569 for its IAX2 communications, especially when it is not acting in a 'server' capacity... so I wouldn't be concerned about that specifically. Given the fact that the phone is not incrementing it's OSeqNo in the REGREQ packets you showed in the capture, I would agree that it appears that the replies from Asterisk are not being received by the phone. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 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
[asterisk-users] Asterisk Queue ACD when the queues and agents has the same priority/weight
Hello, I have three queues (F1,F2,F3) with default queue weight and three agents (A1,A2,A2) with default agent penalty. If the three agents are busy and tt same time a caller (C1) enter in the queue F1, and after 20 seconds a second caller (C2) enter in the queue F2. So, few seconds later, the agent (A1) state comes to availabe. In this case the asterisk deliveries the caller (C2) to agent (A1), but the in the queue (F1) caller (C1) waiting time is bigger compared to caller (C2) of queue (F2). How should be the ACD behavior between queues in this case? How the asterisk distributes incoming calls when the queues and agents are the same weight/penalty? Thanks, - Marcos José Setim -- _ -- 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] Hang using Festival application
Hello, Suppose a dialplan such as: exten = 6004,1,Answer exten = 6004,n,Wait(1) exten = 6004,n,SayDigits(1) exten = 6004,n,Festival(This is a test of Festival) exten = 6004,n,Hangup When watching in the CLI, I see this: == Using SIP RTP CoS mark 5 -- Executing [6004@internal:1] Answer(SIP/505-0004, ) in new stack -- Executing [6004@internal:2] Wait(SIP/505-0004, 1) in new stack -- Executing [6004@internal:3] SayDigits(SIP/505-0004, 1) in new stack -- SIP/505-0004 Playing 'digits/1.gsm' (language 'en') -- Executing [6004@internal:4] Festival(SIP/505-0004, This is a test of Festival) in new stack == Parsing '/etc/asterisk/festival.conf': == Found ps-pbx*CLI ... and nothing more. Nothing happens after == Parsing ..., and the SIP channel gets stuck open even after I physically hang up the extension (will not respond to a hangup request, can only be eliminated by restarting asterisk). I hear one in the phone and then silence. Versions: festival: Festival Speech Synthesis System: 2.0.95:beta April 2010 asterisk: Asterisk 1.6.2.9-2+squeeze1 OS: Debian Squeeze 64 bit ~# uname -a Linux ps-pbx 2.6.32-5-amd64 #1 SMP Wed Jan 12 03:40:32 UTC 2011 x86_64 GNU/Linux These are all unmodified packages obtained via aptitude. What am I getting wrong? Many thanks, ~Brian -- Brian Henning, Software Engineer /\Pine Research Instrumentation //\\ 5908 Triangle Drive ///\\\ Raleigh, NC 27617 USA || ||phone: 919.782.8320 fax: 919.782.8323 email: bhenn...@pineinst.com -- -- _ -- 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] Hang using Festival application
Hi Brian, On Wed, Mar 23, 2011 at 03:15:08PM -0400, Brian Henning wrote: Hello, Suppose a dialplan such as: exten = 6004,1,Answer exten = 6004,n,Wait(1) exten = 6004,n,SayDigits(1) exten = 6004,n,Festival(This is a test of Festival) exten = 6004,n,Hangup ... ... and nothing more. Nothing happens after == Parsing ..., and the SIP channel gets stuck open even after I physically hang up the extension (will not respond to a hangup request, can only be eliminated by restarting asterisk). I hear one in the phone and then silence. ... These are all unmodified packages obtained via aptitude. What am I getting wrong? Your problem might be that you either need to patch Festival or modify the Festival configuration file: http://www.voip-info.org/wiki/view/Asterisk+festival+installation Mark -- Mark G. Thomas (m...@misty.com) Web: http://mgtinternet.com/ Tel: +1-215-512-0112 US: 877-512-0112 -- _ -- 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] using ${EXTEN} with waitexten
All:
Some of the people who dial into to our system will press the pound key
when entering an extension for the directory key. When waitexten gets
that, I get an error messages as, for example 123# doesn't match any
extension.
I was going to use ${EXTEN} to just use the first three numbers, but I'm
not sure how to use this with WaitExten.
so I have
exten = 4349701010,1,Answer()
exten = 4349701010,2,ringing
exten = 4349701010,3,wait(8)
exten = 4349701010,4,Background(asterisk-recording)
exten = 4349701010,5,WaitExten(9,m)
exten = 4349701010,6,Dial(SIP/100SIP/123SIP/132SIP/134SIP/149,20)
exten = 4349701010,7,VoiceMail(100@default,u)
exten = 4349701010,8,Playback(vm-goodbye)
exten = 4349701010,9,Hangup()
Where could I check for the extra # keystroke?
Thanks for your help.
eddie
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[asterisk-users] OT: Have unused DID's; where to warehouse?
We have a set (about 20) of DID's that we're not using. No one calls them, and we don't need them for outgoing. I'd like to keep them for future use. We now pay $5/mo/DID to host them. Is there a way to warehouse them? Just put them in a bank someplace? Thanks, sean -- _ -- 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] dahdi restart warning
What is this error ? is this harmful ? *CLI*CLI dahdi restart [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'userbase' (on reload) at line 23. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'vmsecret' (on reload) at line 31. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hassip' (on reload) at line 35. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hasiax' (on reload) at line 39. [Mar 23 14:01:06] WARNING[4314]: chan_dahdi.c:17422 process_dahdi: Ignoring any changes to 'hasmanager' (on reload) at line 47. [Mar 23 14:01:10] WARNING[4315]: sig_pri.c:985 pri_find_dchan: Span 1: No D-channels available! Using Primary channel as D-channel anyway! -- _ -- 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] using ${EXTEN} with waitexten
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eddie Mikell
Sent: Wednesday, March 23, 2011 3:28 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] using ${EXTEN} with waitexten
All:
Some of the people who dial into to our system will press the pound key
when entering an extension for the directory key. When waitexten gets
that, I get an error messages as, for example 123# doesn't match any
extension.
I was going to use ${EXTEN} to just use the first three numbers, but I'm
not sure how to use this with WaitExten.
so I have
exten = 4349701010,1,Answer()
exten = 4349701010,2,ringing
exten = 4349701010,3,wait(8)
exten = 4349701010,4,Background(asterisk-recording)
exten = 4349701010,5,WaitExten(9,m)
exten = 4349701010,6,Dial(SIP/100SIP/123SIP/132SIP/134SIP/149,20)
exten = 4349701010,7,VoiceMail(100@default,u)
exten = 4349701010,8,Playback(vm-goodbye)
exten = 4349701010,9,Hangup()
Where could I check for the extra # keystroke?
Thanks for your help.
eddie
As I understand it, WaitExten is designed to jump to single-digit extensions
in the same context (at least in 1.4). What you should use here is the Read
command. The output of read is determined by timeout, digit length or #, so
123 and 123# would both evaluate as 123.
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[asterisk-users] asking for some help
hi evrey one, i'm in some kind interesting in developping some asterisk programme like doing a small programme including some of these services that do a telephone operator. but abviously i need to know about programming in asterisk in thos to files i think :) (extensions.conf and in sip.conf files) so i'm asking if someone can give me a puch,i will be very glad thanks in advance -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.8 Packages for Debian and Ubuntu
Greetings, Digium has been providing rpm packages for the latest versions of Asterisk that are compatible with RHEL 5 and CentOS 5 for quite some time now. We are pleased to announce that we will now be providing deb packages for both Debian and Ubuntu. As of now, we have Asterisk 1.8 packages available for the following distribution versions: * Debian 6.0 (squeezy) * Ubuntu 10.04 (lucid) * Ubuntu 10.10 (maverick) * Ubuntu 11.04 (natty) This effort is not intended to replace packaging of Asterisk in the official Debian or Ubuntu repositories. Our repositories are for providing access to major versions of Asterisk that are newer than what is included. We are exploring ways to work as closely as possible with the Debian and Ubuntu package maintainers to ensure that we do not duplicate efforts and that we provide the best possible result for users of Asterisk. For information on how to set up your system to use our repositories, please refer to the following page on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages If you have any problems related to the repositories or the packages themselves, please report them in the AsteriskNOW and Packages project on the Asterisk issue tracker, http://issues.asterisk.org/. Thanks, -- Russell Bryant Digium, Inc. | Engineering Manager, Open Source Software 445 Jan Davis Drive NW- Huntsville, AL 35806 - USA www.digium.com -=- www.asterisk.org -=- blogs.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
[asterisk-users] spa8000 t38 faxing
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Re: [asterisk-users] asking for some help
Hello, I started to work on asterisk 2 years ago. I started from book. I saved it in google docs. You can also start from herehttps://docs.google.com/viewer?a=vpid=explorerchrome=truesrcid=0Bxm7VSlLHvESYmZiMWYyMGUtNDI4OS00NDdjLTkwYjMtZmYxNzM0ZjQ2OGNkhl=enauthkey=CMbXtZMB . On Thu, Mar 24, 2011 at 1:05 AM, tahar .H harazta...@gmail.com wrote: hi evrey one, i'm in some kind interesting in developping some asterisk programme like doing a small programme including some of these services that do a telephone operator. but abviously i need to know about programming in asterisk in thos to files i think :) (extensions.conf and in sip.conf files) so i'm asking if someone can give me a puch,i will be very glad thanks in advance -- _ -- 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 -- Best Regards Shakeel Abbas -- _ -- 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] Forwarding XXXX to XXXX prevented.
I have a Linksys 2102 ATA here that does call forwarding internally with the *72 code, however our Asterisk 1.6.2.17 server doesn't forward the call properly. This is what shows up in the console when an incoming call is made while the ATA is call-forwarded: -- Called Username -- Got SIP response 302 Moved Temporarily back from XX.XXX.XX.XXX -- Now forwarding DAHDI/1-1 to 'Local/12505551234@vancouver' (thanks to SIP/Username-0045) -- Forwarding DAHDI/1-1 to 'Local/12505551234@vancouver' prevented. == Everyone is busy/congested at this time (1:1/0/0) The SIP configuration allows call forwarding (cancallforward=yes), so I'm at a loss as to what is preventing the forwarding. It's not like Asterisk is very specific about that. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT: Have unused DID's; where to warehouse?
At 01:38 PM 3/23/2011, you wrote: I'd like to keep them for future use. We now pay $5/mo/DID to host them. Is there a way to warehouse them? Just put them in a bank someplace? You can pay less then $5. I think I only pay $1.29/DID/month at Flowroute. Ira -- _ -- 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] Issues with Digum Repos / AsteriskNOW Bad Packages
I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error loading module 'app_voicemail_imapstorage.so': /usr/lib/libc-client.so.1: undefined symbol: mm_dlog [Mar 23 23:30:03] WARNING[12592]: loader.c:777 load_resource: Module 'app_voicemail_imapstorage.so' could not be loaded. Is there some way to have this working? -- Med Vennlig Hilsen, A. Helge Joakimsen -- _ -- 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] Problems Extension with a Call In on Asterisk 1.6
Hi Anyone know a solution at my problems ? Thanks Olivier 2011/3/23 Olivier CALVANO o.calv...@gmail.com: Hi I request your help because i don't have actually a solution at my problems. I have a Asterisk Server in 1.6 Connected at a SIP Provider This provider supply me 2 numbers: 003318364 (official number) 081169 (Nddi Number) When i receive a call on the 081169, he don't use the extension. He use the 003318364 extension. SIP Debug: --- SIP read from UDP://91.121.xxx.xxx:5060 --- INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 Allow: UPDATE,REFER,INFO Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Contact: sip:91.121.xxx.xxx:5060 Content-Type: application/sdp CSeq: 1602837515 INVITE From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 P-Preferred-Identity: sip:033426aaa...@sip.myoperator.net;user=phone To: sip:081169x...@91.121.xxx.xxx;user=phone User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 Content-Length: 481 v=0 o=cp10 130085910854 130085910854 IN IP4 10.7.1.121 s=SIP Call c=IN IP4 91.121.bbb.bbb t=0 0 m=audio 36146 RTP/AVP 18 4 0 8 125 111 101 b=AS:21 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000/1 a=fmtp:4 annexa=no a=rtpmap:0 PCMU/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:125 CLEARMODE/8000/1 a=rtpmap:111 iLBC/8000/1 a=fmtp:111 mode=30 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:30 a=sendrecv a=sqn:0 a=cdsc: 1 image udptl t38 - --- (13 headers 22 lines) --- Sending to 91.121.xxx.xxx : 5060 (no NAT) Using INVITE request as basis request - 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Found peer 'Myoperator' for '033426aa' from 91.121.xxx.xxx:5060 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 125 Found RTP audio format 111 Found RTP audio format 101 Peer audio RTP is at port 91.121.bbb.bbb:36146 Found audio description format G729 for ID 18 Found audio description format G723 for ID 4 Found audio description format PCMU for ID 0 Found audio description format PCMA for ID 8 Found unknown media description format CLEARMODE for ID 125 Found audio description format iLBC for ID 111 Found audio description format telephone-event for ID 101 Capabilities: us - 0x109 (g723|alaw|g729), peer - audio=0x50d (g723|ulaw|alaw|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x109 (g723|alaw|g729) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Peer audio RTP is at port 91.121.bbb.bbb:36146 Looking for 003318364 in Appels-Entrants (domain 78.41.xxx.xxx) --- Reliably Transmitting (no NAT) to 91.121.xxx.xxx:5060 --- SIP/2.0 404 Not Found Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2;received=91.121.xxx.xxx From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net CSeq: 1602837515 INVITE Server: Asterisk PBX 1.6.1.8 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527 handle_request_invite: Call from '0033459aa' to extension '003318364' rejected because extension not found. Scheduling destruction of SIP dialog '04459-nk-5fa6f8a0-18641f...@sip.myoperator.net' in 6400 ms (Method: INVITE) --- SIP read from UDP://91.121.xxx.xxx:5060 --- ACK sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0 Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net Contact: sip:91.121.xxx.xxx:5060 CSeq: 1602837515 ACK From: 033426aa sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 To: sip:081169x...@91.121.xxx.xxx;user=phone;tag=as50e04b6a User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 Content-Length: 0 I see in the debug: To: sip:081169x...@91.121.xxx.xxx;user=phone but he search the 003318364 extension [Mar 23 06:45:08] NOTICE[10626]: chan_sip.c:18527 handle_request_invite: Call from '0033459aa' to extension '003318364' rejected because extension not found. Anyone know the solution for he use the extension based on the To: ? thanks Olivier -- _ -- 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:
[asterisk-users] SIP Invite and Asterisk API/Variable
Hi
I have in a SIP invite of a incoming call:
INVITE sip:003318364x...@78.41.xxx.xxx:5060;transport=udp SIP/2.0
Allow: UPDATE,REFER,INFO
Call-ID: 04459-nk-5fa6f8a0-18641f...@sip.myoperator.net
Contact: sip:91.121.xxx.xxx:5060
Content-Type: application/sdp
CSeq: 1602837515 INVITE
From: 033426aa
sip:033426aaa...@sip.myoperator.net;user=phone;tag=04459-CI-5fa6f8a1-6f03b5b60
Max-Forwards: 30
P-Preferred-Identity: sip:033426aaa...@sip.myoperator.net;user=phone
To: sip:081169x...@91.121.xxx.xxx;user=phone
User-Agent: Cirpack/v4.42s (gw_sip)
Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2
Content-Length: 481
The To, To: sip:081169x...@91.121.xxx.xxx;user=phone, can i get it into
a variable for sent it at a API ?
Sample:
in extension.conf:
exten =
_003318364,1,AGI(Caller-ID_Phibee.agi,${CALLERID(name)},${VARIABLE})
exten = _003318364,2,Dial(SIP/185,180,rt)
in this sample, ${VARIABLE} = 081169
Thanks for your help
Olivier
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