Re: [asterisk-users] asterisk with 1000 extensions
2013/3/7 Steve Edwards asterisk@sedwards.com Please don't top-post. On Thu, 7 Mar 2013, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Are you wiring a building where multiple-line SIP gateways make sense? How about a description of what you are trying to do? Personally, I like Polycom SIP phones but I don't have to buy 1,000 of them :) I bet it is a school assignment ... home work or the way you like to call them. However I have a box with 972 peers, no reinvite (but no transcoding), average usage of conference call and other audio mix feature, reaching a max of 60 CPS and an average of 150 channels without problems. The cpu is a double Intel(R) Xeon(R) CPU E5-2630 0 @ 2.30GHz, but it works fine even on the old hardware, a double Intel(R) Xeon(R) CPU 5150 @ 2.66GHz Leandro -- _ -- 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 with 1000 extensions
2013/3/7 Steve Edwards asterisk@sedwards.com Please don't top-post. On Thu, 7 Mar 2013, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Are you wiring a building where multiple-line SIP gateways make sense? How about a description of what you are trying to do? Personally, I like Polycom SIP phones but I don't have to buy 1,000 of them :) I bet it is a school assignment ... home work or the way you like to call them. However I have a box with 972 peers, no reinvite (but no transcoding), average usage of conference call and other audio mix feature, reaching a max of 60 CPS and an average of 150 channels without problems. The cpu is a double Intel(R) Xeon(R) CPU E5-2630 0 @ 2.30GHz, but it works fine even on the old hardware, a double Intel(R) Xeon(R) CPU 5150 @ 2.66GHz Leandro, This is not school assignment or home work :) We need to setup in society buildings. Each flat will have SIP extension (hard phone) registered on asterisk server. Calling between SIP extensions is required. No PSTN / ITSP SIP trunking. Just like inter-com feature. One way is to install 1000 IP Phones one at each flatSecondly, install multiple-line SIP gateways with RJ-11 cabling. Is there any other low budget solution for this setup? Thanks,Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk with 1000 extensions
On 7/03/2013, at 9:29 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: On Thu, 7 Mar 2013, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Are you wiring a building where multiple-line SIP gateways make sense? How about a description of what you are trying to do? Personally, I like Polycom SIP phones but I don't have to buy 1,000 of them :) This is not school assignment or home work :) We need to setup in society buildings. Each flat will have SIP extension (hard phone) registered on asterisk server. Calling between SIP extensions is required. No PSTN / ITSP SIP trunking. Just like inter-com feature. One way is to install 1000 IP Phones one at each flat Secondly, install multiple-line SIP gateways with RJ-11 cabling. Is there any other low budget solution for this setup? Your costs will be in the handsets. Yealink make good cheap phones, you need to find a supplier who can do you a great deal on 1000 phones http://www.yealink.com/product_info.aspx?ProductsCateID=292CateId=147BaseInfoCateId=292Cate_Id=292parentcateid=147?ProductsCateID=292CateId=147BaseInfoCateId=292Cate_Id=292parentcateid=147 But I am not sure why you cant use analogue phones and SIP channel banks such as grandstream or USB ones such as Xorcom. The per line cost will come down and you only need telephony grade cabling to the premise. You can get $10 phones which limit the desire of people to walk off with them The server and setup will cost nothing compared to the handsets Thanks, Kamlesh -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Error to install Asterisk
That should be ok. Try the following: open 2 shells. In the first one type watch df -h. In the second one you start the compilation. While compilation is running watch the first shell. The given command refreshes all 2 seconds the display and shows the used/free disk space. _Perhaps_ it will give you a hint, what mount point is running out of space. Am 06.03.2013 15:41, schrieb termo termosel: I have executed make in the same console where I had written mkdir /var/ext_tmp export TMPDIR=/var/ext_tmp make Is this way ok? Date: Wed, 6 Mar 2013 14:25:50 +0100 From: t...@ovm-group.com To: fermit...@hotmail.com CC: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Error to install Asterisk Did you execute the make command in the same environment so that make really uses the TMPDIR directory? (no su or other shell) Am 06.03.2013 13:37, schrieb termo termosel: Hi, the same error, I write your commands: mkdir /var/ext_tmp export TMPDIR=/var/ext_tmp make but the same error happens /usr/bin/ld: final link failed: No space left on device collect2: ld devolvió el estado de salida 1 make[2]: *** [asterisk] Error 1 make[1]: *** [main] Error 2 make[1]: se sale del directorio «/home/ubuntu/Downloads/asterisk-11.2.1» make: *** [_cleantest_all] Error 2 Jordi Date: Wed, 6 Mar 2013 13:29:24 +0100 From: t...@ovm-group.com mailto:t...@ovm-group.com To: fermit...@hotmail.com mailto:fermit...@hotmail.com CC: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Error to install Asterisk Try to set the tmp variable. In your case: mkdir /var/ext_tmp export TMPDIR=/var/ext_tmp make Am 06.03.2013 13:20, schrieb termo termosel: Hi, I read it but I don't find the solution. How Can I alocate more free space in tmp? Thanks, Jordi Date: Wed, 6 Mar 2013 13:12:34 +0100 From: t...@ovm-group.com mailto:t...@ovm-group.com To: asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com CC: fermit...@hotmail.com mailto:fermit...@hotmail.com Subject: Re: [asterisk-users] Error to install Asterisk Take a look here: http://unix.stackexchange.com/questions/16137/encountering-this-error-usr-bin-ld-final-link-failed-no-space-left-on-device Am 06.03.2013 13:00, schrieb termo termosel: Hi, df -h output: root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# mailto:root@ubuntu:/home/ubuntu/Downloads/asterisk-11.2.1# df -h S.ficherosTam. Usado Disp. % Uso Montado en /cow 14G 4,5G 8,7G 34% / udev 999M 4,0K 999M 1% /dev tmpfs 403M 860K 402M 1% /run /dev/sdb1 799M 693M 106M 87% /cdrom /dev/loop0668M 668M 0 100% /rofs tmpfs1006M 44K 1006M 1% /tmp none 5,0M 0 5,0M 0% /run/lock none 1006M 100K 1006M 1% /run/shm Jordi -- _ -- 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 with 1000 extensions
2013/3/7 Duncan Turnbull dun...@e-simple.co.nz On 7/03/2013, at 9:29 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote: On Thu, 7 Mar 2013, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Are you wiring a building where multiple-line SIP gateways make sense? How about a description of what you are trying to do? Personally, I like Polycom SIP phones but I don't have to buy 1,000 of them :) This is not school assignment or home work :) We need to setup in society buildings. Each flat will have SIP extension (hard phone) registered on asterisk server. Calling between SIP extensions is required. No PSTN / ITSP SIP trunking. Just like inter-com feature. One way is to install 1000 IP Phones one at each flat Secondly, install multiple-line SIP gateways with RJ-11 cabling. Is there any other low budget solution for this setup? Your costs will be in the handsets. Yealink make good cheap phones, you need to find a supplier who can do you a great deal on 1000 phones http://www.yealink.com/product_info.aspx?ProductsCateID=292CateId=147BaseInfoCateId=292Cate_Id=292parentcateid=147?ProductsCateID=292CateId=147BaseInfoCateId=292Cate_Id=292parentcateid=147 But I am not sure why you cant use analogue phones and SIP channel banks such as grandstream or USB ones such as Xorcom. The per line cost will come down and you only need telephony grade cabling to the premise. You can get $10 phones which limit the desire of people to walk off with them The server and setup will cost nothing compared to the handsets Thanks, Kamlesh Sorry, but it is not the first time we help little boys to make homework, it seems asterisk course are common in India and it is easier to cheat than to apply. If you are really trying to serve 1000 phones, beside the usage of SIP or analogue phones via channel banks, I think it will be better to not handle all the load on a single server, but to spread the phone among multiple servers. The best will be to have multiple asterisks working together using realtime extensions. It is not difficult to make. Leandro -- _ -- 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 1.6 + Cisco AS5300
Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 Bad Extension back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-004d == Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-004d' Do you have an explanation? Best regards, Mickael -- _ -- 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 1.6 + Cisco AS5300
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote: Do you have an explanation? Put a SIP debug on and we may be able to find one.. Steve -- _ -- 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 with 1000 extensions
On 7/3/13 6:50 am, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. I would caution against that approach. Analogue to Digital conversions often seem to have 'problems' - mostly related to hangup detection and/or echo. If you really do want to use analogue phones, then use a good quality channel bank to bring the analogue extensions into Asterisk, not low-end ATAs. You also have to consider the value of your time. There's little point shaving a few pounds (or dollars, or euros) from the hardware cost if it's going to double the configuration time. And using 'cheap' components will add to your ongoing support burden for the system. Cheap != good value for money. Personally, I'd consider using something like the Snom 710. They aren't the cheapest SIP phones by any means, but they do have a very good remote provisioning and configuration system, which will substantially reduce the work you need to do in configuring handsets. If your budget won't stretch to the Snom units, the Yealink range as suggested by another poster might be worth looking at. I believe their cheapest (is it the T18?) SIP endpoint can be had for around 35GBP - I don't know what pricing is like in your local currency of course. I believe Yealink do also have a fairly reasonable remote provisioning system, but unlike the Snom system, I can't claim to have used it in anger. Kind regards, Chris -- This email is made from 100% recycled electrons -- _ -- 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 1.6 + Cisco AS5300
Le 7/03/13 11:21, Steven Howes a écrit : On 7 Mar 2013, at 10:12, Mickael Monsieur wrote: Do you have an explanation? Put a SIP debug on and we may be able to find one.. Steve Hello Steve, After checking, I confirm that the call is cut precisely to 900 seconds (15 min). 10.4.0.1 = Asterisk 10.4.0.10 = Cisco AS 5300 Info : debug start at 14min30sec set_destination: Parsing sip:0032487997160@10.4.0.10:5060 for address/port to send to set_destination: set destination to 10.4.0.10, port 5060 Audio is at 10.4.0.1 port 11842 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (NAT) to 10.4.0.10:54789: INVITE sip:0032487997160@10.4.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport Max-Forwards: 70 From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Contact: sip:65939191@10.4.0.1 Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 INVITE User-Agent: isdnbox1.1 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 207 v=0 o=root 1538728127 1538728127 IN IP4 10.4.0.1 s=Asterisk PBX 1.6.2.9-2+squeeze8 c=IN IP4 10.4.0.1 t=0 0 m=audio 11842 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- --- SIP read from UDP:10.4.0.10:5060 --- SIP/2.0 420 Bad Extension Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 INVITE Unsupported: timer Content-Length: 0 - --- (8 headers 0 lines) --- -- Got SIP response 420 Bad Extension back from 10.4.0.10 set_destination: Parsing sip:0032487997160@10.4.0.10:5060 for address/port to send to set_destination: set destination to 10.4.0.10, port 5060 Transmitting (NAT) to 10.4.0.10:5060: ACK sip:0032487997160@10.4.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport Max-Forwards: 70 From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Contact: sip:65939191@10.4.0.1 Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 ACK User-Agent: isdnbox1.1 Content-Length: 0 --- -- Stopped music on hold on SIP/as5300-1-0050 == Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-0050' Reliably Transmitting (NAT) to 10.4.0.10:5060: OPTIONS sip:10.4.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport Max-Forwards: 70 From: asterisk sip:asterisk@10.4.0.1;tag=as4eb3efa7 To: sip:10.4.0.10 Contact: sip:asterisk@10.4.0.1 Call-ID: 6a43ad4b27d870d048e8425077bcc075@10.4.0.1 CSeq: 102 OPTIONS User-Agent: isdnbox1.1 Date: Thu, 07 Mar 2013 11:17:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- --- SIP read from UDP:10.4.0.10:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport From: asterisk sip:asterisk@10.4.0.1;tag=as4eb3efa7 To: sip:10.4.0.10;tag=37A724C-211C Date: Sat, 01 Jan 2000 16:12:32 GMT Call-ID: 6a43ad4b27d870d048e8425077bcc075@10.4.0.1 Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp CSeq: 102 OPTIONS Supported: 100rel Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Accept: application/sdp Allow-Events: telephone-event Content-Length: 154 v=0 o=CiscoSystemsSIP-GW-UserAgent 793 5073 IN IP4 10.4.0.10 s=SIP Call c=IN IP4 10.4.0.10 t=0 0 m=audio 0 RTP/AVP 18 0 8 4 2 15 3 c=IN IP4 10.4.0.10 - --- (14 headers 7 lines) --- Really destroying SIP dialog '6a43ad4b27d870d048e8425077bcc075@10.4.0.1' Method: OPTIONS --- SIP read from UDP:10.4.0.10:54336 --- BYE sip:65939191@10.4.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.10:5060 From: sip:0032487997160@10.4.0.10;tag=36CA05C-167B To: sip:65939191@10.4.0.1;tag=as12acaefb Date: Sat, 01 Jan 2000 16:12:26 GMT Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 6 Timestamp: 946743153 CSeq: 102 BYE Content-Length: 0 - --- (11 headers 0 lines) --- --- Transmitting (NAT) to 10.4.0.10:54336 --- SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.4.0.10:5060;received=10.4.0.10 From: sip:0032487997160@10.4.0.10;tag=36CA05C-167B To: sip:65939191@10.4.0.1;tag=as12acaefb Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 BYE Server: isdnbox1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 15 min (call ended) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com
Re: [asterisk-users] asterisk with 1000 extensions
From: Chris Bagnall aster...@lists.minotaur.cc To: asterisk-users@lists.digium.com, Date: 03/07/2013 06:43 AM Subject:Re: [asterisk-users] asterisk with 1000 extensions Sent by:asterisk-users-boun...@lists.digium.com On 7/3/13 6:50 am, Bharat Lalcheta wrote: You can use ATA box with pstn phone to reduce cost. Cheap != good value for money. I am going to go with Chris on this. Don't look for the absolute cheapest, look for the absolute best value, balancing the cost of the end units versus the cost spent to support them. We use Polycom phones and while they are not the cheapest, I never have to mess with them other than to update configs to add in some new feature I am working on. Rock solid, even after 5 years and many firmware upgrades. -- _ -- 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] 11.3: how to hang up on google voice
Some calls I get from google voice, I just send myself an email about
the call and want to hangup. But I can't seem to make gv know I've hung up.
extensions.conf:
same = n,GoToIf($[${CALLERID(num)}=office]?email)
.
same = n(email),System(/usr/local/bin/emailme)
same = n,Answer() ; also tried without this
same = n,Hangup()
console:
-- Executing [s@incoming-171:13] Answer(Motif/+1...-1b35, )
in new stack
-- Executing [s@incoming-171:14] Hangup(Motif/+1..-1b35,
) in new stack
== Spawn extension (incoming-171, s, 14) exited non-zero on
'Motif/+1..-1b35'
but the calling phone keeps ringing until the google voice attendant
comes on.
OTOH, if I dial an extension, hanging up works fine.
So how do I get gv to recognize a hangup?
sean
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Re: [asterisk-users] 11.3: how to hang up on google voice
sean darcy wrote:
Some calls I get from google voice, I just send myself an email about
the call and want to hangup. But I can't seem to make gv know I've hung up.
extensions.conf:
same = n,GoToIf($[${CALLERID(num)}=office]?email)
.
same = n(email),System(/usr/local/bin/emailme)
same = n,Answer() ; also tried without this
same = n,Hangup()
You need to Answer, Wait, send a DTMF of 1, wait a bit more, and then
hang up.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: www.digium.com www.asterisk.org
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Re: [asterisk-users] Asterisk 1.6 + Cisco AS5300
Can u debug on AS ? On Thu, Mar 7, 2013 at 9:20 AM, Mickael Monsieur mickael.monsi...@gmail.com wrote: Le 7/03/13 11:21, Steven Howes a écrit : On 7 Mar 2013, at 10:12, Mickael Monsieur wrote: Do you have an explanation? Put a SIP debug on and we may be able to find one.. Steve Hello Steve, After checking, I confirm that the call is cut precisely to 900 seconds (15 min). 10.4.0.1 = Asterisk 10.4.0.10 = Cisco AS 5300 Info : debug start at 14min30sec set_destination: Parsing sip:0032487997160@10.4.0.10:5060 for address/port to send to set_destination: set destination to 10.4.0.10, port 5060 Audio is at 10.4.0.1 port 11842 Adding codec 0x8 (alaw) to SDP Adding codec 0x4 (ulaw) to SDP Reliably Transmitting (NAT) to 10.4.0.10:54789: INVITE sip:0032487997160@10.4.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport Max-Forwards: 70 From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Contact: sip:65939191@10.4.0.1 Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 INVITE User-Agent: isdnbox1.1 Require: timer Session-Expires: 1800;refresher=uas Min-SE: 90 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer X-asterisk-Info: SIP re-invite (Session-Timers) Content-Type: application/sdp Content-Length: 207 v=0 o=root 1538728127 1538728127 IN IP4 10.4.0.1 s=Asterisk PBX 1.6.2.9-2+squeeze8 c=IN IP4 10.4.0.1 t=0 0 m=audio 11842 RTP/AVP 8 0 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv --- --- SIP read from UDP:10.4.0.10:5060 --- SIP/2.0 420 Bad Extension Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 INVITE Unsupported: timer Content-Length: 0 - --- (8 headers 0 lines) --- -- Got SIP response 420 Bad Extension back from 10.4.0.10 set_destination: Parsing sip:0032487997160@10.4.0.10:5060 for address/port to send to set_destination: set destination to 10.4.0.10, port 5060 Transmitting (NAT) to 10.4.0.10:5060: ACK sip:0032487997160@10.4.0.10:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK41af21b3;rport Max-Forwards: 70 From: sip:65939191@10.4.0.1;tag=as12acaefb To: sip:0032487997160@10.4.0.10;tag=36CA05C-167B Contact: sip:65939191@10.4.0.1 Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 ACK User-Agent: isdnbox1.1 Content-Length: 0 --- -- Stopped music on hold on SIP/as5300-1-0050 == Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-0050' Reliably Transmitting (NAT) to 10.4.0.10:5060: OPTIONS sip:10.4.0.10 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport Max-Forwards: 70 From: asterisk sip:asterisk@10.4.0.1;tag=as4eb3efa7 To: sip:10.4.0.10 Contact: sip:asterisk@10.4.0.1 Call-ID: 6a43ad4b27d870d048e8425077bcc075@10.4.0.1 CSeq: 102 OPTIONS User-Agent: isdnbox1.1 Date: Thu, 07 Mar 2013 11:17:44 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 --- --- SIP read from UDP:10.4.0.10:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 10.4.0.1:5060;branch=z9hG4bK4d8b5654;rport From: asterisk sip:asterisk@10.4.0.1;tag=as4eb3efa7 To: sip:10.4.0.10;tag=37A724C-211C Date: Sat, 01 Jan 2000 16:12:32 GMT Call-ID: 6a43ad4b27d870d048e8425077bcc075@10.4.0.1 Server: Cisco-SIPGateway/IOS-12.x Content-Type: application/sdp CSeq: 102 OPTIONS Supported: 100rel Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO Accept: application/sdp Allow-Events: telephone-event Content-Length: 154 v=0 o=CiscoSystemsSIP-GW-UserAgent 793 5073 IN IP4 10.4.0.10 s=SIP Call c=IN IP4 10.4.0.10 t=0 0 m=audio 0 RTP/AVP 18 0 8 4 2 15 3 c=IN IP4 10.4.0.10 - --- (14 headers 7 lines) --- Really destroying SIP dialog '6a43ad4b27d870d048e8425077bcc075@10.4.0.1' Method: OPTIONS --- SIP read from UDP:10.4.0.10:54336 --- BYE sip:65939191@10.4.0.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.0.10:5060 From: sip:0032487997160@10.4.0.10;tag=36CA05C-167B To: sip:65939191@10.4.0.1;tag=as12acaefb Date: Sat, 01 Jan 2000 16:12:26 GMT Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 User-Agent: Cisco-SIPGateway/IOS-12.x Max-Forwards: 6 Timestamp: 946743153 CSeq: 102 BYE Content-Length: 0 - --- (11 headers 0 lines) --- --- Transmitting (NAT) to 10.4.0.10:54336 --- SIP/2.0 481 Call leg/transaction does not exist Via: SIP/2.0/UDP 10.4.0.10:5060;received=10.4.0.10 From: sip:0032487997160@10.4.0.10;tag=36CA05C-167B To: sip:65939191@10.4.0.1;tag=as12acaefb Call-ID: FA122D28-BF9A11D3-83D393BC-25F3EF03@10.4.0.10 CSeq: 102 BYE Server: isdnbox1.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE,
Re: [asterisk-users] Asterisk 1.6 + Cisco AS5300
Le 7/03/13 11:12, Mickael Monsieur a écrit : Hello, I have a Cisco AS5300 connected to Asterisk (1.6.2.9) Between 15-16 minutes, the call is disconnected without reason. Here is what is displayed in the debug: Received an SDES from 10.4.0.10:17399 -- Got SIP response 420 Bad Extension back from 10.4.0.10 -- Stopped music on hold on SIP/as5300-1-004d == Spawn extension (dialin, 065939191, 2) exited non-zero on 'SIP/as5300-1-004d' Do you have an explanation? Best regards, Mickael Ok i solved : https://issues.asterisk.org/jira/browse/ASTERISK-15787 -- _ -- 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] 11.3: how to hang up on google voice
There might be something wrong with the evaluation. Can you post more
console regarding the GotoIf?
And are you sure its in the right context?
Sent from my iPhone
On 7 mrt. 2013, at 15:48, sean darcy seandar...@gmail.com wrote:
Some calls I get from google voice, I just send myself an email about the
call and want to hangup. But I can't seem to make gv know I've hung up.
extensions.conf:
same = n,GoToIf($[${CALLERID(num)}=office]?email)
.
same = n(email),System(/usr/local/bin/emailme)
same = n,Answer() ; also tried without this
same = n,Hangup()
console:
-- Executing [s@incoming-171:13] Answer(Motif/+1...-1b35, ) in new
stack
-- Executing [s@incoming-171:14] Hangup(Motif/+1..-1b35, ) in
new stack
== Spawn extension (incoming-171, s, 14) exited non-zero on
'Motif/+1..-1b35'
but the calling phone keeps ringing until the google voice attendant comes on.
OTOH, if I dial an extension, hanging up works fine.
So how do I get gv to recognize a hangup?
sean
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[asterisk-users] Recording with MixMonitor and AGI
Hi,
I am developing a call recording application on Asterisk 11.2 and have this
configuration in my dialplan:
[macro-ccdev2-rec]
exten = s,1,MixMonitor(${ARG1},b)
[outgoing-originate]
exten = _X.,1,NoOp(Will send call to ${EXTEN})
exten = _X.,n,Dial(SIP/${EXTEN}@x.y.z)
[outgoing-originate-rec]
exten = h,1,Agi(agi://localhost/ajpbx.agi?path=uploadreccallid=${CC_CALLID})
exten = _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is ${CC_CALLID},
CC_FILENAME is ${CC_FILENAME})
exten = _X,n,Dial(SIP/${EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e)
If I want to make a recorded server callout from 0 to 08 I then
originate a call via AMI to Local/0@outgoing-originate with context set
to outgoing-originate-rec and extension to 08.
The result will be something like this:
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-003f
-- Executing [h@outgoing-originate-rec:1] AGI(SIP/upps-ccm-tq01-003e,
agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack
-- SIP/upps-ccm-tq01-003eAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed, returning 0
-- Executing [h@outgoing-originate-rec-dev2:1]
AGI(SIP/upps-ccm-tq01-003f,
agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack
-- SIP/upps-ccm-tq01-003fAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed, returning 0
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/upps-ccm-tq01-003f
Unfortunately I get two different calls to the h extension, but this I can cope
with. The one without called is not interesting.
The uploading will fail since the MixMonitor is still on when I try to upload
the file. The file will not have a duration. It works when I schedule the
uploading a while after from my agi application but I would rather not rely on
a timeout.
When I tried to run StopMixMonitor before the Agi call in the h extension, the
first call fail and I never get any uploading with callid.
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-0043, cbrec-15607.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-0043
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0042, ) in new stack
== Spawn extension (outgoing-originate-rec-dev2, h, 1) exited non-zero on
'SIP/upps-ccm-tq01-0042'
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0043, ) in new stack
== MixMonitor close filestream (mixed)
-- Executing [h@outgoing-originate-rec-dev2:2]
AGI(SIP/upps-ccm-tq01-0043,
agi://localhost/ajpbxtest.agi?path=uploadreccallid=) in new stack
Am I missing something here? I also looked at the possibility to specify a
command to execute when MixMonitor stops but I would rather handle the file
uploading in my agi application.
I also have another case: I want to dial out a call and record it. It will be a
oneway-call from the server to a mobile. Do I need to get AGI-control of it
and record with an AGI command or how can I hack it directly in the dial plan
using MixMonitor?
Best Regards,
Henrik
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Re: [asterisk-users] 11.3: how to hang up on google voice
On 03/07/2013 09:48 AM, Joshua Colp wrote:
sean darcy wrote:
Some calls I get from google voice, I just send myself an email about
the call and want to hangup. But I can't seem to make gv know I've
hung up.
extensions.conf:
same = n,GoToIf($[${CALLERID(num)}=office]?email)
.
same = n(email),System(/usr/local/bin/emailme)
same = n,Answer() ; also tried without this
same = n,Hangup()
You need to Answer, Wait, send a DTMF of 1, wait a bit more, and then
hang up.
Brilliant.
same = n(hangup),Answer()
same = n,Wait(3)
same = n,SendDTMF(1)
same = n,Wait(3)
same = n,Hangup()
Worked like a charm. It does cause gv to give a circuit busy. But that's ok.
sean
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Re: [asterisk-users] asterisk with 1000 extensions
On Thu, Mar 7, 2013 at 1:44 AM, Duncan Turnbull dun...@e-simple.co.nzwrote: This is not school assignment or home work :) We need to setup in society buildings. Each flat will have SIP extension (hard phone) registered on asterisk server. Calling between SIP extensions is required. No PSTN / ITSP SIP trunking. Just like inter-com feature. One way is to install 1000 IP Phones one at each flat Secondly, install multiple-line SIP gateways with RJ-11 cabling. Is there any other low budget solution for this setup? Grandstream makes some inexpensive phones that are still very good. Cheapest hasn't been defined yet. What's the budget? Is there existing networking at these locations? Will you need switches? PoE? -- Carlos Alvarez TelEvolve 602-889-3003 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Extension cant pickup calls but can transfer.
Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* *** -- _ -- 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] Extension cant pickup calls but can transfer.
If I was in your shoes, I'll check in the elastix mailing list... Asterisk itself can't be blamed. Leandro I am typing from my mobile phone... Il giorno 07/mar/2013 19:06, Luis H. Forchesatto luisforchesa...@gmail.com ha scritto: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* *** -- _ -- 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] Extension cant pickup calls but can transfer.
On 8/03/2013, at 7:46 AM, Leandro Dardini ldard...@gmail.com wrote: If I was in your shoes, I'll check in the elastix mailing list... Asterisk itself can't be blamed. Leandro I am typing from my mobile phone... Il giorno 07/mar/2013 19:06, Luis H. Forchesatto luisforchesa...@gmail.com ha scritto: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. More often than not you haven't set the pickup group correctly in the extension config in freepbx Otherwise something in the phone might be stealing the code and using it for something else. Where should I look? Any further information needed just ask. -- Att. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crashed
On 03/07/2013 10:32 AM, Zohair Raza wrote: Its Centos 6 with kernel 2.6.32-279.19.1.el6.x86_64 Please follow the instructions on the wiki [1] for generating a backtrace. When you have a backtrace from the crash, please create an issue in the issue tracker [2] and attach the backtrace to it. Thanks! [1] https://wiki.asterisk.org/wiki/display/AST/Getting+a+Backtrace [2] https://issues.asterisk.org/jira -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk crashed
Did u test it without abrt? On Mar 7, 2013 10:03 PM, Zohair Raza engineerzuhairr...@gmail.com wrote: Its Centos 6 with kernel 2.6.32-279.19.1.el6.x86_64 Regards, Zohair Raza On Thu, Mar 7, 2013 at 8:28 AM, Bharat Lalcheta bharatlalch...@gmail.comwrote: Can you provide OS details ? Its seems problem of abrt. Did u tested asterisk without abrt Regards, Bharat Lalcheta On Thu, Mar 7, 2013 at 12:05 AM, Zohair Raza engineerzuhairr...@gmail.com wrote: Hi, I am running asterisk 1.8.14.0, It was running fine for last few days and suddenly crashed today In logs I can see that abrt tried to save the core dump but it couldn't Mar 6 12:11:09 localhost kernel: asterisk[26544]: segfault at 72656d69ac ip 00533c19 sp 7f7db9ce3af0 error 4 in asterisk[40+1d1000] Mar 6 12:11:15 localhost abrt[31287]: Saved core dump of pid 26528 (/usr/sbin/asterisk) to /var/spool/abrt/ccpp-2013-03-06-12:11:09-26528 (450703360 bytes) Mar 6 12:11:15 localhost abrtd: Directory 'ccpp-2013-03-06-12 :11:09-26528' creation detected Mar 6 12:11:15 localhost abrtd: Executable '/usr/sbin/asterisk' doesn't belong to any package Mar 6 12:11:15 localhost abrtd: 'post-create' on '/var/spool/abrt/ccpp-2013-03-06-12:11:09-26528' exited with 1 *Asterisk was running as root user Any suggestions? Regards, Zohair Raza -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bharat Lalcheta -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Recording with MixMonitor and AGI
hi,
hard to understand, what your objective is... at least for me ;-)
so you want to establish a call (triggered by ami) between two partys,
record the conversation
and save the file to a(nother) server (afterwards), right?
and another task is to establish (also ami triggered) a call to a mobile
and play, lets say a voicefile.
this conversation should also be recorded and saved on a(nother)
server (afterwards), right?
let me know, if i understood you right, the solution is not so hard to
implement.
In what language do you preferrably write your AGIs? (although there is
no absolute need for using an
agi... you can all write down in your dialplan...)
is there a special protocol requirement for saving/transferring the
recorded voicefile (e.g. ftps)?
One obstacle is, that the recorded file is not fully written
_immediately_ after stopmixmonitor or hangup...
this has to be taken care of and depending on your agi... it might be
interrupted, if the call is hungup...
but as you did not show your agi... these are just hints..
regards,
yves
Am 07.03.2013 16:21, schrieb Henrik Westerberg:
Hi,
I am developing a call recording application on Asterisk 11.2 and have
this configuration in my dialplan:
[macro-ccdev2-rec]
exten = s,1,MixMonitor(${ARG1},b)
[outgoing-originate]
exten = _X.,1,NoOp(Will send call to ${EXTEN})
exten = _X.,n,Dial(SIP/${EXTEN}@x.y.z)
[outgoing-originate-rec]
exten =
h,1,Agi(agi://localhost/ajpbx.agi?path=uploadreccallid=${CC_CALLID})
exten = _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is
${CC_CALLID}, CC_FILENAME is ${CC_FILENAME})
exten = _X,n,Dial(SIP/${EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e)
If I want to make a recorded server callout from 0
to 08 I then originate a call via AMI to
Local/0@outgoing-originate with context set
to outgoing-originate-rec and extension to 08.
The result will be something like this:
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-003f
-- Executing [h@outgoing-originate-rec:1]
AGI(SIP/upps-ccm-tq01-003e,
agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack
-- SIP/upps-ccm-tq01-003eAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed,
returning 0
-- Executing [h@outgoing-originate-rec-dev2:1]
AGI(SIP/upps-ccm-tq01-003f,
agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack
-- SIP/upps-ccm-tq01-003fAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed,
returning 0
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/upps-ccm-tq01-003f
Unfortunately I get two different calls to the h extension, but this I
can cope with. The one without called is not interesting.
The uploading will fail since the MixMonitor is still on when I try to
upload the file. The file will not have a duration. It works when I
schedule the uploading a while after from my agi application but I
would rather not rely on a timeout.
When I tried to run StopMixMonitor before the Agi call in the h
extension, the first call fail and I never get any uploading with callid.
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-0043, cbrec-15607.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-0043
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0042, ) in new stack
== Spawn extension (outgoing-originate-rec-dev2, h, 1) exited
non-zero on 'SIP/upps-ccm-tq01-0042'
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0043, ) in new stack
== MixMonitor close filestream (mixed)
-- Executing [h@outgoing-originate-rec-dev2:2]
AGI(SIP/upps-ccm-tq01-0043,
agi://localhost/ajpbxtest.agi?path=uploadreccallid=) in new stack
Am I missing something here? I also looked at the possibility to
specify a command to execute when MixMonitor stops but I would rather
handle the file uploading in my agi application.
I also have another case: I want to dial out a call and record it. It
will be a oneway-call from the server to a mobile. Do I need to get
AGI-control of it and record with an AGI command or how can I hack it
directly in the dial plan using MixMonitor?
Best Regards,
Henrik
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Re: [asterisk-users] Extension cant pickup calls but can transfer.
do you have only ONE phone, that can´t pickup, or is this a general problem? is pickup configured (feature.conf) AND enabled ? regards, yves Am 07.03.2013 19:05, schrieb Luis H. Forchesatto: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* * -- _ -- 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] Extension cant pickup calls but can transfer.
Its only ONE phone who doesnt pickup calls. 2013/3/7 Yves A. yves...@gmx.de do you have only ONE phone, that can´t pickup, or is this a general problem? is pickup configured (feature.conf) AND enabled ? regards, yves Am 07.03.2013 19:05, schrieb Luis H. Forchesatto: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* * -- _ -- 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 -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.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] Extension cant pickup calls but can transfer.
is it the same type and make of phone than one of the working ones? - compare (dtmf) settings, firmware release etc. - check call-group and pickup group... is the non working extension configured there? regards, yves Am 07.03.2013 20:28, schrieb Luis H. Forchesatto: Its only ONE phone who doesnt pickup calls. 2013/3/7 Yves A. yves...@gmx.de mailto:yves...@gmx.de do you have only ONE phone, that can´t pickup, or is this a general problem? is pickup configured (feature.conf) AND enabled ? regards, yves Am 07.03.2013 19:05, schrieb Luis H. Forchesatto: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* * -- _ -- Bandwidth and Colocation Provided byhttp://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 -- Att.* * Luis H. Forchesatto Mail: luis_forchesa...@hotmail.com mailto:luis_forchesa...@hotmail.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 -- _ -- 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] Extension cant pickup calls but can transfer.
Yes, both are configured in the same ata (linksys pap2) and the configuration options are the same. Call group and pick group are the same for both too. 2013/3/7 Yves A. yves...@gmx.de is it the same type and make of phone than one of the working ones? - compare (dtmf) settings, firmware release etc. - check call-group and pickup group... is the non working extension configured there? regards, yves Am 07.03.2013 20:28, schrieb Luis H. Forchesatto: Its only ONE phone who doesnt pickup calls. 2013/3/7 Yves A. yves...@gmx.de do you have only ONE phone, that can´t pickup, or is this a general problem? is pickup configured (feature.conf) AND enabled ? regards, yves Am 07.03.2013 19:05, schrieb Luis H. Forchesatto: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* * -- _ -- 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 -- Att.* * Luis H. Forchesatto Mail: luis_forchesa...@hotmail.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 -- _ -- 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 -- Att.* *** Luis H. Forchesatto Mail: luis_forchesa...@hotmail.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] Extension cant pickup calls but can transfer.
mmh... should work... (i think you checked double and applied any changes, right..? sometimes deleting the extension and configuring a new one can fulfil wonders...) I have no further tip... maybe elastix support or forum can help... if you are familiar with cli output and sip debugging... check cli output and sip debug output... good luck. yves Am 07.03.2013 20:38, schrieb Luis H. Forchesatto: Yes, both are configured in the same ata (linksys pap2) and the configuration options are the same. Call group and pick group are the same for both too. 2013/3/7 Yves A. yves...@gmx.de mailto:yves...@gmx.de is it the same type and make of phone than one of the working ones? - compare (dtmf) settings, firmware release etc. - check call-group and pickup group... is the non working extension configured there? regards, yves Am 07.03.2013 20:28, schrieb Luis H. Forchesatto: Its only ONE phone who doesnt pickup calls. 2013/3/7 Yves A. yves...@gmx.de mailto:yves...@gmx.de do you have only ONE phone, that can´t pickup, or is this a general problem? is pickup configured (feature.conf) AND enabled ? regards, yves Am 07.03.2013 19:05, schrieb Luis H. Forchesatto: Greetings. I got an extension on my Elastix who cannot pick calls on the other extensions, but It can transfer his calls to the other extensions. When this extension tries to pickup a call pressing *8 it simply does not pick it up. Transfering calls works just fine so dtmf may be not the problem. Where should I look? Any further information needed just ask. -- Att.* * -- _ -- Bandwidth and Colocation Provided byhttp://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 -- Att.* * Luis H. Forchesatto Mail: luis_forchesa...@hotmail.com mailto:luis_forchesa...@hotmail.com -- _ -- Bandwidth and Colocation Provided byhttp://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 -- Att.* * Luis H. Forchesatto Mail: luis_forchesa...@hotmail.com mailto:luis_forchesa...@hotmail.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 -- _ -- 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] Ring back issue with asterisk 1.8.18.0
Hi, Here is the configuration of the server that I currently have extension 100 (SIP) =(SIP)asterisk server 1.8.18(IAX trunk) ===(IAX trunk)asterisk server 1.4.32(SIP) === SIP Providers The issue is while dialing out from extension 100(sip) if the providers sends back 180 Rining the SIP extension(100) won't hear the ringback tone, where as if the providers send 183 session in progress extension(100) will hear the ring back tone. I tested registering to the main gateway server (by passing iax trunk) and it plays ring back tone every time for 180 Rining and for 183 session progress. Any help would be highly appreciated. Thanks -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] AGI Script
Hi, thanks. .Did that include 'agi set debug on?' Yes, that include this command. Can you 'cut-n-paste' the relevant 'sanitized' console output? Ok. This is: trixbox146002*CLI -- Launched AGI Script /var/lib/asterisk/agi-bin/agi-test.agi trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_request: agi-test.agi SIP/INCONCERT-35d2AGI Tx agi_channel: SIP/INCONCERT-35d2 trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_language: en trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_type: SIP trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_uniqueid: 1362669295.35561 trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_version: 1.6.0.28-samy-r100 SIP/INCONCERT-35d2AGI Tx agi_callerid: 1044 trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_calleridname: 1044 trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_callingpres: 0 trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_callingani2: 0 trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_callington: 0 trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_callingtns: 0 trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_dnid: 701820101044 trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_rdnis: unknown trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_context: ModernaDesv_E1 trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_extension: s trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_priority: 3 trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_enhanced: 0.0 trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_accountcode: trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx agi_threadid: 29367216 trixbox146002*CLI SIP/INCONCERT-35d2AGI Tx trixbox146002*CLI [Mar 7 10:14:55] NOTICE[32548]: channel.c:3051 __ast_read: Dropping incompatible voice frame on SIP/INCONCERT-35d2 of format ulaw since our native format has changed to 0x8 (alaw) trixbox146002*CLI -- SIP/INCONCERT-35d2AGI Script agi-test.agi completed, returning 0 I looked through my AGIs and find I always set channel variables and let the dialplan do the actual dial(). 1) Is your AGI exiting before the dial() completes? Yes. 2) If you execute the same dial() command from the 'AGI debug output' (which should show the expanded variables) in your dialplan, does that yield andy clues? How I do that? I just use the AMI command originate. Executing this with the protocol already works, but not ring. 3) If you use another technology like SIP can you enable SIP debugging and observe the SIP dialog? I can not use SIP because the call goes out to E1. Regards, Gustavo -- _ -- 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] Polycom SPIP config
Has any one ever worked with placing idle display images onto the Polycom SPIP331 phones? I have got it working but when the image is displayed the clock is moved to the top of the screen. That is great but it scrolls between the clock and the registered extension(s) . Has anyone figured out a way to stop the scrolling and just display the time? If so could you provide me the configuration parameter? thanks, Bryan Anderson -- _ -- 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] VOIP PRI Gateways
I was hoping someone might have some knowledge to impart regarding VOIP PRI Gateways or the psudo ISDN services being offered these days. The official line in Australia is that true ISDN services are on their way out. I am testing a service provided by one of the telcos I am told that it cannot provide ISDN cause codes for disconnected/invalid numbers and all we get is the audio. Also the service no longer sends the progress event RINGING to indicate the line is actual ringing (as apposed to the audio) Has anyone else ran into this problem and does anyone have any ideas how to address it? Can DAHDI detect ringing tones on PRI lines? -- Cheers, Daniel -- _ -- 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] Polycom SPIP config
On Thu, 7 Mar 2013 17:12:47 -0800 Bryan Anderson shadow...@gmail.com wrote: Has any one ever worked with placing idle display images onto the Polycom SPIP331 phones? I have got it working but when the image is displayed the clock is moved to the top of the screen. That is great but it scrolls between the clock and the registered extension(s) . Has anyone figured out a way to stop the scrolling and just display the time? If so could you provide me the configuration parameter? Sorry to say... we have the same problem with the 321s. Never managed to figure it out. I asked Polycom about it, and they said we'd have to get our vendor to order it as a feature request, or something like that. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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] Recording with MixMonitor and AGI
As far as i understand your requirements, there is no need to use
macro for recording, You can directly call mixmonitor before Dial
application in your dialplan with required options. For transfer of
file, you are using AGI in h priority. However, you have to use
DeadAgi in h extension. As your channel already hangup, it can not
run on AGI.
Hope it will help you.
Regards,
Bharat Lalcheta
On Thu, Mar 7, 2013 at 8:51 PM, Henrik Westerberg
henrik.westerb...@ain.se wrote:
Hi,
I am developing a call recording application on Asterisk 11.2 and have this
configuration in my dialplan:
[macro-ccdev2-rec]
exten = s,1,MixMonitor(${ARG1},b)
[outgoing-originate]
exten = _X.,1,NoOp(Will send call to ${EXTEN})
exten = _X.,n,Dial(SIP/${EXTEN}@x.y.z)
[outgoing-originate-rec]
exten =
h,1,Agi(agi://localhost/ajpbx.agi?path=uploadreccallid=${CC_CALLID})
exten = _X,1,NoOp(Will send call to ${EXTEN}, CC_CALLID is ${CC_CALLID},
CC_FILENAME is ${CC_FILENAME})
exten = _X,n,Dial(SIP/${EXTEN}@x.y.z,60,M(ccdev2-rec^${CC_FILENAME})e)
If I want to make a recorded server callout from 0 to 08 I
then originate a call via AMI to Local/0@outgoing-originate with
context set to outgoing-originate-rec and extension to 08.
The result will be something like this:
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-003f, cbrec-15605.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-003f
-- Executing [h@outgoing-originate-rec:1]
AGI(SIP/upps-ccm-tq01-003e,
agi://l4574/ajpbxtest.agi?path=uploadreccallid=15605) in new stack
-- SIP/upps-ccm-tq01-003eAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid=15605 completed,
returning 0
-- Executing [h@outgoing-originate-rec-dev2:1]
AGI(SIP/upps-ccm-tq01-003f,
agi://4574/ajpbxtest.agi?path=uploadreccallid=) in new stack
-- SIP/upps-ccm-tq01-003fAGI Script
agi://localhost/ajpbxtest.agi?path=uploadreccallid= completed, returning 0
== MixMonitor close filestream (mixed)
== End MixMonitor Recording SIP/upps-ccm-tq01-003f
Unfortunately I get two different calls to the h extension, but this I can
cope with. The one without called is not interesting.
The uploading will fail since the MixMonitor is still on when I try to
upload the file. The file will not have a duration. It works when I schedule
the uploading a while after from my agi application but I would rather not
rely on a timeout.
When I tried to run StopMixMonitor before the Agi call in the h extension,
the first call fail and I never get any uploading with callid.
-- Executing [s@macro-ccdev2-rec:1]
MixMonitor(SIP/upps-ccm-tq01-0043, cbrec-15607.wav,b) in new stack
== Begin MixMonitor Recording SIP/upps-ccm-tq01-0043
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0042, ) in new stack
== Spawn extension (outgoing-originate-rec-dev2, h, 1) exited non-zero on
'SIP/upps-ccm-tq01-0042'
-- Executing [h@outgoing-originate-rec-dev2:1]
StopMixMonitor(SIP/upps-ccm-tq01-0043, ) in new stack
== MixMonitor close filestream (mixed)
-- Executing [h@outgoing-originate-rec-dev2:2]
AGI(SIP/upps-ccm-tq01-0043,
agi://localhost/ajpbxtest.agi?path=uploadreccallid=) in new stack
Am I missing something here? I also looked at the possibility to specify a
command to execute when MixMonitor stops but I would rather handle the file
uploading in my agi application.
I also have another case: I want to dial out a call and record it. It will
be a oneway-call from the server to a mobile. Do I need to get AGI-control
of it and record with an AGI command or how can I hack it directly in the
dial plan using MixMonitor?
Best Regards,
Henrik
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Re: [asterisk-users] VOIP PRI Gateways
If I understand you correctly, you test a service which converter SIP to ISDN PRI On Mar 8, 2013, at 2:16 AM, Daniel Harper dan...@harper.net.nz wrote: I was hoping someone might have some knowledge to impart regarding VOIP PRI Gateways or the psudo ISDN services being offered these days. The official line in Australia is that true ISDN services are on their way out. I am testing a service provided by one of the telcos I am told that it cannot provide ISDN cause codes for disconnected/invalid numbers and all we get is the audio. Also the service no longer sends the progress event RINGING to indicate the line is actual ringing (as apposed to the audio) Has anyone else ran into this problem and does anyone have any ideas how to address it? Can DAHDI detect ringing tones on PRI lines? -- Cheers, Daniel -- _ -- 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
