[asterisk-users] How to notify that a message is waiting using RingTones with a Dahdi-connected analog phone ?
Hi, I'm wondering how I could notify to a dumb analog phone that a voicemail message is waiting. My goal would be to change the tone that is heard just before user starts to dial. Any idea on that ? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk-GUI
On Mon, Jul 06, 2009 at 11:19:49AM +0900, Tseveendorj wrote: Hello, I've installed Asterisk 1.4.21.2 with Asterisk-GUI on Ubuntu 9.04 but Asterisk. I couldn't see asterisk-GUI web interface when I accessed to http://IPADDRESS:8088/asterisk/static/config/index.html. But after created symbolic link /var/lib/asterisk/static-http folder to /usr/share/asterisk/ and /var/lib/asterisk/scripts to /usr/share/asterisk/ web interface appeared. Why doesn't asterisk read static-http from /var/lib/asterisk/ ? https://issues.asterisk.org/view.php?id=15119 -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP registry fails during night
Every morning I check my SIP registry to the SIP-provider. And I must conclude that during the night somewhere registry has failed. asterisk*CLI sip show registry HostUsername Refresh State Reg.Time 85.119.188.3:5060 092779077 105 Failed Sun, 05 Jul 2009 23:11:40 asterisk*CLI sip reload [Jul 6 10:30:43] Reloading SIP [Jul 6 10:30:43] == Parsing '/etc/asterisk/sip.conf': [Jul 6 10:30:43] Found [Jul 6 10:30:43] == Parsing '/etc/asterisk/users.conf': [Jul 6 10:30:43] Found [Jul 6 10:30:43] == Parsing '/etc/asterisk/sip_notify.conf': [Jul 6 10:30:43] Found asterisk*CLI sip show registry HostUsername Refresh State Reg.Time 85.119.188.3:5060 092779077 105 Registered Mon, 06 Jul 2009 10:30:43 I must do a 'sip reload' to get registered again. What could be failing ?? Is this a NAT issue of some kind ? Could it be that my firewall at one point blocks things off ??? If it has something to do with NAT or firewall, why does a simple 'sip reload' gets me registered again ?! Greetingz, Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best way to share extension state
asterisk-us...@rogg.is writes: 3) Hook into the AMI and parse the ExtensionStatusEvent which I think gives me what I want. This is the traditional way, I believe. The challenge with AMI is that it is becoming a high-bandwidth channel. If you're only interested in one event type, you will spend quite a bit of CPU time just discarding uninteresting events. Perhaps it would be possible to make events more granular, instead of just on/off? Apart from that AMI works fine. /Benny ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP registry fails during night
On 6 Jul 2009, at 09:37, jonas kellens wrote: What could be failing ?? Is this a NAT issue of some kind ? Could it be that my firewall at one point blocks things off ??? If it has something to do with NAT or firewall, why does a simple 'sip reload' gets me registered again ?! Don't use a Netgear by any chance do you? S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP registry fails during night
Thanks for your reply, Steve. My firewall is a 3-NIC pc with Endian Community installed. Jonas. On Mon, 2009-07-06 at 09:49 +0100, Steve Howes wrote: On 6 Jul 2009, at 09:37, jonas kellens wrote: What could be failing ?? Is this a NAT issue of some kind ? Could it be that my firewall at one point blocks things off ??? If it has something to do with NAT or firewall, why does a simple 'sip reload' gets me registered again ?! Don't use a Netgear by any chance do you? S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP registry fails during night
Hi, Hmm. Not used that. Just out of interest, do you have 'qualify' on the peer in Asterisk? Could it be that its all a bit quiet and your router is breaking the stateful firewall stuff? Steve On 6 Jul 2009, at 10:05, jonas kellens wrote: Thanks for your reply, Steve. My firewall is a 3-NIC pc with Endian Community installed. Jonas. On Mon, 2009-07-06 at 09:49 +0100, Steve Howes wrote: On 6 Jul 2009, at 09:37, jonas kellens wrote: What could be failing ?? Is this a NAT issue of some kind ? Could it be that my firewall at one point blocks things off ??? If it has something to do with NAT or firewall, why does a simple 'sip reload' gets me registered again ?! Don't use a Netgear by any chance do you? S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk capacity
Hello, This is the configuration of my server got from PHP system info: *System Vital:* Kernel Version 2.6.18-6-amd64 (SMP) Distro Name Debian 4.0 *Hardware Information:* Processors 4 Model Intel(R) Xeon(R) CPU E5420 @ 2.50GHz CPU Speed 2.49 GHz Cache Size 6.00 MB System Bogomips 19954.78 PCI Devices *none* IDE Devices *none* SCSI Devices -DELL PERC 6/i (Direct-Access)-DP BACKPLANE (Enclosure)-TSSTcorp DVD-ROM TS-L333A (CD-ROM) USB Devices -Dell Computer Corp.-Cypress Semiconductor Corp. CY7C65640 USB-2.0 TetraHub-Dell Computer Corp. Hub* Mounted Filesystems:* *Mount* *Type* *Partition* *Percent Capacity* *Free* * Used* *Size* / ext3 /dev/sda1 0% 759.28 GB 1.63 GB 801.63 GB /dev/shm tmpfs tmpfs 0% (1%) 3.90 GB 0.00 KB 3.90 GB /lib/init/rw tmpfs tmpfs 0% (1%) 3.90 GB 0.00 KB 3.90 GB /dev tmpfs udev 1% (1%) 9.95 MB 52.00 KB 10.00 MB *Totals : * 0% 763.19 GB 1.63 GB 805.54 GB * ** Memory Usage:* *Type* *Percent Capacity* *Free* *Used* *Size* Physical Memory 5% 7.37 GB 437.23 MB 7.80 GB - Kernel + applications 2% 194.45 MB- Buffers 2% 159.57 MB- Cached 1% 83.21 MBDisk Swap 0% 22.84 GB 0.00 KB 22.84 GB * The version of Asterisk is: 1.4.22. I need to know how many calls I can handle with my Asterisk. Thks. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] asterisk and mISDN on Solaris
Hi, I read that installing asterisk on Solaris is supported. Does anyone of you actually have experiences with that? And especially, does anyone of you have experiences in runnning asterisk with misdn unter Solaris? Thanks and regards, Christophorus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP registry fails during night
Hi Steve, I have qualify for all the peers that are defined, and so also for the peer I have defined for my SIP-provider. What you see below is such qualify : [Jul 6 11:38:26] Reliably Transmitting (NAT) to 85.119.188.3:5060: OPTIONS sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK0299a153;rport From: asterisk sip:aster...@192.168.2.2;tag=as2c85bba5 To: sip:85.119.188.3 Contact: sip:aster...@192.168.2.2 Call-ID: 4561f356290f4f9a06b4b5bb10250...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Mon, 06 Jul 2009 09:38:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jul 6 11:38:26] --- SIP read from 85.119.188.3:5060 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK0299a153;rport=5060 From: asterisk sip:aster...@192.168.2.2:5060;tag=as2c85bba5 To: sip:85.119.188.3;tag=470f6df9d438ef0e611aed43a3c90fcf.2711 Call-ID: 4561f356290f4f9a06b4b5bb10250...@192.168.2.2 CSeq: 102 OPTIONS Server: Enswitch SIP proxy Content-Length: 0 Warning: 392 85.119.188.3:5060 Noisy feedback tells: pid=7147 req_src_ip=78.22.164.52 req_src_port=5060 in_uri=sip:85.119.188.3 out_uri=sip:85.119.188.3 via_cnt==1 Don't know what to think about the warning on the last line... Qualify is defined to check every minute... qualify=yes; defaults to 60 seconds Registration is defined to try for 1 hour, and then give up : registertimeout=60 ; retry registration calls every 20 seconds (default) registerattempts=60; Number of registration attempts before we give up ; 0 = continue forever What do you mean when you say that my router is breaking stateful traffic ?? It is also my firewall that is doing the NAT-ing. Traffic that is initiated from the DMZ (like a SIP option from my Asterisk) and that is replied to within 1 second should not be an issue for a stateful firewall, is it ?! Jonas. On Mon, 2009-07-06 at 10:15 +0100, Steve Howes wrote: Hi, Hmm. Not used that. Just out of interest, do you have 'qualify' on the peer in Asterisk? Could it be that its all a bit quiet and your router is breaking the stateful firewall stuff? Steve ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Monitor
Hi All
am using trixbox with call queues..I've set setinterfacevars=yes in queues.conf
and following is dial plan :
[test]
exten = s,1,Answer()
exten =
s,2,Set(FILE_NAME=${CALLERID(num)}-${MEMBERINTERFACE}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = s,3,Monitor(wav,${FILE_NAME},m)
exten = s,4,queue(55365)
exten = s,5,Hangup()
but MEMBERINTERFACE is always empty - i basically want to add the member who
took that call in that monitor file..i tried in trixbox too bt problem
persists...can anyone throw some light ?
rgds
Sriram___
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[asterisk-users] Iax trunk quality
Hi, I try to find a solution for this problem : [Jul 3 09:30:38] WARNING[3756]: chan_iax2.c:7312 socket_process: Received trunked frame before first full voice frame [Jul 3 09:30:38] WARNING[3757]: chan_iax2.c:7312 socket_process: Received trunked frame before first full voice frame [Jul 3 09:30:38] WARNING[3751]: chan_iax2.c:7312 socket_process: Received trunked frame before first full voice frame [Jul 3 09:30:38] WARNING[3759]: chan_iax2.c:7312 socket_process: Received trunked frame before first full voice frame [Jul 3 09:30:38] WARNING[3752]: chan_iax2.c:7312 socket_process: Received trunked frame before first full voice frame ...etc ... In the same time the user complain of the telephony quality. I have 14 ms of ping in my trunk. And i use Cisco with QOS on a dedicaced SDSL. *CLI iax2 show netstats LOCAL - REMOTE Channel RTT Jit Del Lost % Drop OOO Kpkts Jit Del Lost % Drop OOO Kpkts IAX2/blabla-16385 15 -1 0 -1 -1 0 -1 0 0 40 0 0 0 0 0 IAX2/blabla-16387 12 -1 0 -1 -1 0 -1 0 0 40 0 0 0 0 0 2 active IAX channels ## My configuration ( I use a sangoma key for the timing source on my slave.) : server_master:/etc/asterisk# asterisk -r Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. server_slave:/etc/asterisk# asterisk -r Asterisk 1.4.24.1, Copyright (C) 1999 - 2008 Digium, Inc. and others. server_master:/var/lib/asterisk/agi-bin/inc# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.998535% 99.995308% 99.996284% 99.997269% 99.998924% 99.996872% 99.998634% 99.993553% 99.998245% 99.995216% 99.998039% 99.997856% 99.992966% 99.999321% --- Results after 14 passes --- Best: 99.999 -- Worst: 99.993 -- Average: 99.996930, Difference: 99.997334 ; iax.conf server_master ; [general] bindport= bindaddr=222.222.222.222 delayreject=yes language=fr allow=alaw maxjitterbuffer=800 trunktimestamps=yes tos=ef [jourdain] type=friend host=111.111.111.111 port= context=iax-jourdain trunk=yes disallow=all allow=alaw jitterbuffer=no forcejitterbuffer=no transfer=no server_salve:~# dahdi_test Opened pseudo dahdi interface, measuring accuracy... 99.989647% 99.985260% 99.989449% 99.989357% 99.989555% 99.989456% 99.989853% 99.989548% 99.986908% 99.989258% 99.989250% 99.989250% 99.990234% 99.989647% ^C --- Results after 14 passes --- Best: 99.990 -- Worst: 99.985 -- Average: 99.989048, Difference: 99.989048 server_slave*CLI iax2 show peers Name/Username Host Mask Port Status toulouse 222.222.222.222 (S) 255.255.255.255 (T) OK (18 ms) ; ; iax.conf server_salve ; [general] bindport= bindaddr=111.111.111.111 language=fr maxjitterbuffer=250 trunktimestamps=yes tos=ef [toulouse] type=friend host=222.222.222.222 port= context=iax-toulouse trunk=yes allow=alaw jitterbuffer=no forcejitterbuffer=no transfer=no Best regards Hugues ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Monitor
if you haven't exectued the queue cmd you cannot know who will took that call.
You cannot know this before the agent took it because there are many agents who
can do it.
You can know it via cdr or manager interface, but only when the call is tooked
or finished.
On Mon, Jul 06, 2009 at 03:23:29PM +0530, Sriram wrote:
Hi All
am using trixbox with call queues..I've set setinterfacevars=yes in
queues.conf and following is dial plan :
[test]
exten = s,1,Answer()
exten =
s,2,Set(FILE_NAME=${CALLERID(num)}-${MEMBERINTERFACE}-${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
exten = s,3,Monitor(wav,${FILE_NAME},m)
exten = s,4,queue(55365)
exten = s,5,Hangup()
but MEMBERINTERFACE is always empty - i basically want to add the member who
took that call in that monitor file..i tried in trixbox too bt problem
persists...can anyone throw some light ?
rgds
Sriram
___
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--
Roger Casaponsa - Adam Telefonía IP
email: roger.casapo...@adamvozip.es mailto:roger.casapo...@adamvozip.es
www: http://www.adamvozip.es http://www.adamvozip.es/
tlf: 902546800
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Re: [asterisk-users] Grandstream 2010 and blinky lights
Thanks for the info. We've managed to achieve or goal using 1.4 and a few hacks. 1) When the agent logs in / logs out, we rewrite the part of the dialplan for the hints and reload the dialplan 10 seconds after the *last* login / logout 2) For the MWI, we give each phone a fake voicemail (let's say _0001_). When an agent logs in, we link /var/spool/asterisk/voicemail/_0001_ to /var/spool/asterisk/voicemail/[mailbox] (where [mailbox] is the mailbox of the agent) and when they log out, we remove /var/spool/asterisk/voicemail/_0001_ This seems to work - the MWI lights up / off depending on the new vm within a couple of seconds 3) When checking for voicemail, each phone is configured to dial - the dialplan then checks the callerid (set by #1) and gets the mailbox for the agent. As I said, a bit of a hack, but it works for me ;) I know that this won't work for 1.6, but we are coming up with an alternative plan using Minivm Julian Andrew Thomas wrote: The quick answer is 'no'. It is not currently possible to monitor 'hints' for Agents - as an Agent never actually dials out (the device does). Even exten = 1234,hint,Agent/1234 won't work - as the 'core show hints' will show the agent as 'notinuse' when they can be. There are ways around it (I used a mixture of php and mysql) - but even these are not ideal (especially if you have a large dial plan). Clue : exten 1234,hint,SIP/ABC works - you just need to change the ABC bit every time an agent logs in our out. This then gives you the lovely job of lighting any MWI lamps for that user as well. Oh the joys of Asterisk and hotdesking! HTH Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 02 July 2009 17:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Grandstream 2010 and blinky lights I am using 1.4, and have the above device, and it worked really well with monitoring 18 hints aka devices. Now, I've moved us to a hotdesking paradigm where the user is the extension not the device. IOW if I dial 1234, I will get user 1234 (who happens to log on to device ABC today, and DEF tomorrow). Can I make the GXP monitor user 1234, not extension 1234 ? Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Grandstream 2010 and blinky lights
The quick answer is 'no'. It is not currently possible to monitor 'hints' for Agents - as an Agent never actually dials out (the device does). Even exten = 1234,hint,Agent/1234 won't work - as the 'core show hints' will show the agent as 'notinuse' when they can be. There are ways around it (I used a mixture of php and mysql) - but even these are not ideal (especially if you have a large dial plan). Clue : exten 1234,hint,SIP/ABC works - you just need to change the ABC bit every time an agent logs in our out. This then gives you the lovely job of lighting any MWI lamps for that user as well. Oh the joys of Asterisk and hotdesking! HTH Andrew Thomas Technical Services Manager DataVox Ltd Saddleworth Business Centre Huddersfield Road Delph, Oldham OL3 5DF -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Julian Lyndon-Smith Sent: 02 July 2009 17:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Grandstream 2010 and blinky lights I am using 1.4, and have the above device, and it worked really well with monitoring 18 hints aka devices. Now, I've moved us to a hotdesking paradigm where the user is the extension not the device. IOW if I dial 1234, I will get user 1234 (who happens to log on to device ABC today, and DEF tomorrow). Can I make the GXP monitor user 1234, not extension 1234 ? Julian __ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email __ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP registry fails during night
Registration timed out... This time a 'sip reload' doesn't help (why would it?). Registration uses no NAT, and a SIP option uses NAT... Verbosity is at least 25 [Jul 6 11:58:09] -- Remote UNIX connection [Jul 6 11:58:39] NOTICE[30045]: chan_sip.c:7683 sip_reg_timeout:-- Registration for '092779...@85.119.188.3' timed out, trying again (Attempt #9) [Jul 6 11:59:00] WARNING[30045]: chan_sip.c:1980 retrans_pkt: Maximum retries exceeded on transmission 50bc585d1d8a5bdc1e2d17250e935...@127.0.0.1 for seqno 196 (Critical Request) -- See doc/sip-retransmit.txt. [Jul 6 11:59:40] NOTICE[30045]: chan_sip.c:7683 sip_reg_timeout:-- Registration for '092779...@85.119.188.3' timed out, trying again (Attempt #10) [Jul 6 12:00:00] WARNING[30045]: chan_sip.c:1980 retrans_pkt: Maximum retries exceeded on transmission 50bc585d1d8a5bdc1e2d17250e935...@127.0.0.1 for seqno 197 (Critical Request) -- See doc/sip-retransmit.txt. [Jul 6 12:00:40] NOTICE[30045]: chan_sip.c:7683 sip_reg_timeout:-- Registration for '092779...@85.119.188.3' timed out, trying again (Attempt #11) [Jul 6 12:01:00] WARNING[30045]: chan_sip.c:1980 retrans_pkt: Maximum retries exceeded on transmission 50bc585d1d8a5bdc1e2d17250e935...@127.0.0.1 for seqno 198 (Critical Request) -- See doc/sip-retransmit.txt. [Jul 6 12:01:40] NOTICE[30045]: chan_sip.c:7683 sip_reg_timeout:-- Registration for '092779...@85.119.188.3' timed out, trying again (Attempt #12) [Jul 6 12:02:00] WARNING[30045]: chan_sip.c:1980 retrans_pkt: Maximum retries exceeded on transmission 50bc585d1d8a5bdc1e2d17250e935...@127.0.0.1 for seqno 199 (Critical Request) -- See doc/sip-retransmit.txt. asterisk*CLI sip reload [Jul 6 12:02:02] Reloading SIP [Jul 6 12:02:02] == Parsing '/etc/asterisk/sip.conf': [Jul 6 12:02:02] Found [Jul 6 12:02:02] == Parsing '/etc/asterisk/users.conf': [Jul 6 12:02:02] Found [Jul 6 12:02:02] == Parsing '/etc/asterisk/sip_notify.conf': [Jul 6 12:02:02] Found [Jul 6 12:02:22] WARNING[30045]: chan_sip.c:1980 retrans_pkt: Maximum retries exceeded on transmission 6ae9293718f8b7d8272b711e45b79...@127.0.0.1 for seqno 102 (Critical Request) -- See doc/sip-retransmit.txt. [Jul 6 12:03:02] NOTICE[30045]: chan_sip.c:7683 sip_reg_timeout:-- Registration for '092779...@85.119.188.3' timed out, trying again (Attempt #1) asterisk*CLI sip show registry HostUsername Refresh State Reg.Time 85.119.188.3:5060 092779077 120 Request Sent [Jul 6 12:08:15] Retransmitting #5 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK2d03d319;rport From: sip:092779...@85.119.188.3;tag=as2ea4597b To: sip:092779...@85.119.188.3 Call-ID: 6ae9293718f8b7d8272b711e45b79...@127.0.0.1 CSeq: 108 REGISTER User-Agent: Asterisk-jocan Max-Forwards: 70 Expires: 120 Contact: sip:s...@192.168.2.2 Event: registration Content-Length: 0 --- [Jul 6 12:08:19] Retransmitting #6 (no NAT) to 85.119.188.3:5060: REGISTER sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK2d03d319;rport From: sip:092779...@85.119.188.3;tag=as2ea4597b To: sip:092779...@85.119.188.3 Call-ID: 6ae9293718f8b7d8272b711e45b79...@127.0.0.1 CSeq: 108 REGISTER User-Agent: Asterisk-jocan Max-Forwards: 70 Expires: 120 Contact: sip:s...@192.168.2.2 Event: registration Content-Length: 0 --- [Jul 6 12:10:29] Retransmitting #2 (NAT) to 85.119.188.3:5060: OPTIONS sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK1e0c1c91;rport From: asterisk sip:aster...@192.168.2.2;tag=as5ef29e25 To: sip:85.119.188.3 Contact: sip:aster...@192.168.2.2 Call-ID: 04ddff5e283ab4767a3bc30a2d31f...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Mon, 06 Jul 2009 10:10:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 --- [Jul 6 12:10:30] Retransmitting #3 (NAT) to 85.119.188.3:5060: OPTIONS sip:85.119.188.3 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.2:5060;branch=z9hG4bK1e0c1c91;rport From: asterisk sip:aster...@192.168.2.2;tag=as5ef29e25 To: sip:85.119.188.3 Contact: sip:aster...@192.168.2.2 Call-ID: 04ddff5e283ab4767a3bc30a2d31f...@192.168.2.2 CSeq: 102 OPTIONS User-Agent: Asterisk-jocan Max-Forwards: 70 Date: Mon, 06 Jul 2009 10:10:27 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk capacity
It can make 9977.39 Bogocalls of course! On Mon, Jul 6, 2009 at 5:17 AM, abdelkaderabdelkader2...@gmail.com wrote: Hello, This is the configuration of my server got from PHP system info: System Vital: Kernel Version 2.6.18-6-amd64 (SMP) Distro Name Debian 4.0 Hardware Information: Processors 4 Model Intel(R) Xeon(R) CPU E5420 @ 2.50GHz CPU Speed 2.49 GHz Cache Size 6.00 MB System Bogomips 19954.78 PCI Devices none IDE Devices none SCSI Devices -DELL PERC 6/i (Direct-Access) -DP BACKPLANE (Enclosure) -TSSTcorp DVD-ROM TS-L333A (CD-ROM) USB Devices -Dell Computer Corp. -Cypress Semiconductor Corp. CY7C65640 USB-2.0 TetraHub -Dell Computer Corp. Hub Mounted Filesystems: Mount Type Partition Percent Capacity Free Used Size / ext3 /dev/sda1 0% 759.28 GB 1.63 GB 801.63 GB /dev/shm tmpfs tmpfs 0% (1%) 3.90 GB 0.00 KB 3.90 GB /lib/init/rw tmpfs tmpfs 0% (1%) 3.90 GB 0.00 KB 3.90 GB /dev tmpfs udev 1% (1%) 9.95 MB 52.00 KB 10.00 MB Totals : 0% 763.19 GB 1.63 GB 805.54 GB Memory Usage: Type Percent Capacity Free Used Size Physical Memory 5% 7.37 GB 437.23 MB 7.80 GB - Kernel + applications 2% 194.45 MB - Buffers 2% 159.57 MB - Cached 1% 83.21 MB Disk Swap 0% 22.84 GB 0.00 KB 22.84 GB The version of Asterisk is: 1.4.22. I need to know how many calls I can handle with my Asterisk. Thks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best way to share extension state
Benny Amorsen schrieb: asterisk-us...@rogg.is writes: 3) Hook into the AMI and parse the ExtensionStatusEvent which I think gives me what I want. The challenge with AMI is that it is becoming a high-bandwidth channel. If you're only interested in one event type, you will spend quite a bit of CPU time just discarding uninteresting events. Perhaps it would be possible to make events more granular, instead of just on/off? After you're logged in send Action: Events EventMask: call or Action: Events EventMask: call,system if you're interested in reload etc. But you are right. call still gives you events like NewExten, NewChannel etc. apart from ExtensionStatus. NewExten can be pretty verbose. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk capacity
You can handle 600 SIP sessions and about 400 calls doing transcoding ( passing RTP ) On Mon, Jul 6, 2009 at 1:26 PM, Steve Totaro stot...@totarotechnologies.com wrote: It can make 9977.39 Bogocalls of course! On Mon, Jul 6, 2009 at 5:17 AM, abdelkaderabdelkader2...@gmail.com wrote: Hello, This is the configuration of my server got from PHP system info: System Vital: Kernel Version 2.6.18-6-amd64 (SMP) Distro Name Debian 4.0 Hardware Information: Processors 4 Model Intel(R) Xeon(R) CPU E5420 @ 2.50GHz CPU Speed 2.49 GHz Cache Size 6.00 MB System Bogomips 19954.78 PCI Devices none IDE Devices none SCSI Devices -DELL PERC 6/i (Direct-Access) -DP BACKPLANE (Enclosure) -TSSTcorp DVD-ROM TS-L333A (CD-ROM) USB Devices -Dell Computer Corp. -Cypress Semiconductor Corp. CY7C65640 USB-2.0 TetraHub -Dell Computer Corp. Hub Mounted Filesystems: Mount Type Partition Percent Capacity Free Used Size / ext3 /dev/sda1 0% 759.28 GB 1.63 GB 801.63 GB /dev/shm tmpfs tmpfs 0% (1%) 3.90 GB 0.00 KB 3.90 GB /lib/init/rw tmpfs tmpfs 0% (1%) 3.90 GB 0.00 KB 3.90 GB /dev tmpfs udev 1% (1%) 9.95 MB 52.00 KB 10.00 MB Totals :0% 763.19 GB 1.63 GB 805.54 GB Memory Usage: Type Percent Capacity Free Used Size Physical Memory 5% 7.37 GB 437.23 MB 7.80 GB - Kernel + applications 2% 194.45 MB - Buffers 2% 159.57 MB - Cached 1% 83.21 MB Disk Swap 0% 22.84 GB 0.00 KB 22.84 GB The version of Asterisk is: 1.4.22. I need to know how many calls I can handle with my Asterisk. Thks. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk capacity
Steve Totaro schrieb: It can make 9977.39 Bogocalls of course! Mind to share the formula? Wait. Got it. Bogomips/2. Why on earth isn't that documented?! ;) On Mon, Jul 6, 2009 at 5:17 AM, abdelkaderabdelkader2...@gmail.com wrote: Kernel Version 2.6.18-6-amd64 (SMP) Distro Name Debian 4.0 Hardware Information: Processors 4 Model Intel(R) Xeon(R) CPU E5420 @ 2.50GHz CPU Speed 2.49 GHz Cache Size 6.00 MB System Bogomips 19954.78 PCI Devices none IDE Devices none SCSI Devices -DELL PERC 6/i (Direct-Access) -DP BACKPLANE (Enclosure) -TSSTcorp DVD-ROM TS-L333A (CD-ROM) USB Devices -Dell Computer Corp. -Cypress Semiconductor Corp. CY7C65640 USB-2.0 TetraHub -Dell Computer Corp. Hub Mounted Filesystems: Mount Type Partition Percent Capacity Free Used Size / ext3 /dev/sda1 0% 759.28 GB 1.63 GB 801.63 GB /dev/shm tmpfs tmpfs 0% (1%) 3.90 GB 0.00 KB 3.90 GB /lib/init/rw tmpfs tmpfs 0% (1%) 3.90 GB 0.00 KB 3.90 GB /dev tmpfs udev 1% (1%) 9.95 MB 52.00 KB 10.00 MB Totals :0% 763.19 GB 1.63 GB 805.54 GB Memory Usage: Type Percent Capacity Free Used Size Physical Memory 5% 7.37 GB 437.23 MB 7.80 GB - Kernel + applications 2% 194.45 MB - Buffers 2% 159.57 MB - Cached 1% 83.21 MB Disk Swap 0% 22.84 GB 0.00 KB 22.84 GB Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best way to share extension state
2009/7/6 asterisk-us...@rogg.is Greetings. I wonder what is the best way in your opinion to share real-time extension state with applications outside of asterisk? What do you exactly mean by applications ? Do you mean a single server application or several instances of client applications ? ... Sincerely, Baldvin ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queues recording CDR
Hello,
Just a correction, Asternic Call Center Stats is not from
asteriskguru. Asteriskguru has its own statistic program that is not
open source, but free to use. Asternic was written by me (not
asteriskguru) and has an open source version and a commercial one.
Best regards,
--
Nicolás Gudiño
Buenos Aires - Argentina
On Mon, Jul 6, 2009 at 12:08 AM, Kurian Thayilkurianmtha...@gmail.com wrote:
Hi Sriram,
1. Set the channel variable MonitorFilename before Queue() in dialplan
and you can give some meaningful filename for record.
2. I guess you can use an AGI to capture events and then integrate this
with a DB in the Backend. This should help you to track the activity.
3. asternic from asteriskguru is kind of OK. Gives you a live and
detailed report. Parses the queue_log to the MySQL DB and works. This
parse program could be used in your AGI which I mentioned in point 2.
Hope this helps.
Regards,
Kurian Thayil.
On Sun, 2009-07-05 at 22:41 +0530, Sriram wrote:
Hi
1. I want to record all calls that land to an agent via a queue using
a meaningful name - as of now i name the recorded file on the fly
using {CALLERID} variable so that the file gets stored using the
caller id iunder /var/spool/asterisk/monitor , now if i want to store
it as CALLERIDEXTEN where call landed from queue how can i do
this ?
2. I have a CDR issue - when A calls he is put in Queue and say he is
answered by Agent B ..Agent B transfers the Call to agent C as it is
to Agent C whom A wants to talk..when the call gets d/c the CDR for
that call shows the destination field as B whereas it shd be C...how
do i take care of this ...in my call center agents are paid on the
basis of talk time on inbound calls - this way an agent who just
transfers calls is at merry !!
3. Are their any GPL based queue reporting software - hows the
asterisk queue statistics program from asteriskguru.com has anyone
tried it ?
Thanks
Sriram
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Kurian Mathew Thayil.
(GPG KeyID: E232394F)
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Re: [asterisk-users] SIP registry fails during night
Hi! Every morning I check my SIP registry to the SIP-provider. And I must conclude that during the night somewhere registry has failed. I must do a 'sip reload' to get registered again. Can you ALWAYS solve this with a SIP RELOAD, or is it sometimes necessary to restart Asterisk? Anyway, take a look: https://issues.asterisk.org/view.php?id=15052 https://issues.asterisk.org/view.php?id=15139 https://issues.asterisk.org/view.php?id=14518 https://issues.asterisk.org/view.php?id=12312 Maybe these bugs will be of help to you - try to change the register address to use the IP address instead of hostname and see if that improves the situation. Philipp ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to notify that a message is waiting using RingTones with a Dahdi-connected analog phone ?
Olivier wrote: Hi, I'm wondering how I could notify to a dumb analog phone that a voicemail message is waiting. My goal would be to change the tone that is heard just before user starts to dial. Any idea on that ? Yea, it's called stutter dial tone. For DAHDI channels just specify the mailbox in chan_dahdi.conf. If it's connected to an ATA then specify the mailbox on the peer in sip.conf/iax.conf. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem configuring TDM400
jonas kellens wrote: On Fri, 2009-07-03 at 11:58 +0100, Mike wrote: tempest:~# lspci 00:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I don't think this is you TDM-card... This is mine : 04:05.0 Ethernet controller: Digium, Inc. TDM400P (rev 11) Subsystem: Digium, Inc. TDM400P Flags: bus master, medium devsel, latency 32, IRQ 90 I/O ports at d100 [size=256] Memory at ff74 (32-bit, non-prefetchable) [size=1K] Expansion ROM at 8000 [disabled] [size=128K] Capabilities: [c0] Power Management version 2 I don't think your XEN VM can see your TDM-card. You will need to ad a module to your XEN-kernel to be able to speak to your TDM pci-card. Don't know if this module exists... My TDM400P appears as his does: 00:14.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error
I have installed gnutls and gnutls-devel from RedHat repositories [r...@asterisk asterisk]# yum install gnutls gnutls-devel I have installed iksemel with gnutls support : [r...@asterisk asterisk]# cd /usr/src/iksemel-1.3/ [r...@asterisk asterisk]# ./configure --with-gnutls --prefix=/usr [r...@asterisk asterisk]# make [r...@asterisk asterisk]# make check [r...@asterisk asterisk]# make install [r...@asterisk asterisk]# ls -l /usr/lib | grep iksemel -rw-r--r-- 1 root root 184210 2009-07-06 14:52 libiksemel.a -rwxr-xr-x 1 root root 816 2009-07-06 14:52 libiksemel.la lrwxrwxrwx 1 root root 19 2009-07-06 14:52 libiksemel.so - libiksemel.so.3.1.0 lrwxrwxrwx 1 root root 19 2009-07-06 14:52 libiksemel.so.3 - libiksemel.so.3.1.0 -rwxr-xr-x 1 root root 138938 2009-07-06 14:52 libiksemel.so.3.1.0 Then compiled Asterisk again : [r...@asterisk asterisk]# cd /usr/src/asterisk-1.4.25.1/ [r...@asterisk asterisk]# make clean [r...@asterisk asterisk]# ./configure [r...@asterisk asterisk]# make menuconfig [r...@asterisk asterisk]# make [r...@asterisk asterisk]# make install Then edited jabber.conf : [general] debug=yes ;;Turn on debugging by default. autoprune=no;;Auto remove users from buddy list. autoregister=yes;;Auto register users from buddy list. [asterisk] ;;label type=component ;;Client or Component connection serverhost=192.168.2.5 ;;Route to server for example talk.google.com username=aster...@192.168.2.5 ;;Username with optional roster. secret=XX ;;Password port=5222 ;;Port to use defaults to 5222 usetls=yes ;;Use tls or not ;usesasl=yes;;Use sasl or not statusmessage=I am Asterisk ;;Have custom status message for Asterisk. ;timeout=100;;Timeout on the message stack. Then start Asterisk : [r...@asterisk asterisk]# /usr/sbin/asterisk -c And this is the error concerning jabber when wanting to connect to my OpenFire-server: [Jul 6 15:15:36] JABBER: reconnecting. [Jul 6 15:15:36] JABBER: asterisk OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:component:accept' to='aster...@192.168.2.5' version='1.0' [Jul 6 15:15:36] JABBER: asterisk INCOMING: ?xml version='1.0' encoding='UTF-8'?stream:stream from=openfire.jocan.local id=7pI2f xmlns=jabber:component:accept xmlns:stream=http://etherx.jabber.org/streams; version=1.0stream:error xmlns:stream=http://etherx.jabber.org/streams;bad-namespace-prefix xmlns=urn:ietf:params:xml:ns:xmpp-streams//stream:error [Jul 6 15:15:36] JABBER: asterisk OUTGOING: handshake2313234e99edf2891db7901990cf854e8e5639c3/handshake [Jul 6 15:15:36] JABBER: asterisk INCOMING: /stream:stream [Jul 6 15:15:40] WARNING[23732]: res_jabber.c:1573 aji_recv_loop: JABBER: socket read error [Jul 6 15:15:40] JABBER: reconnecting. [Jul 6 15:15:40] JABBER: asterisk OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:component:accept' to='aster...@192.168.2.5' version='1.0' [Jul 6 15:15:40] JABBER: asterisk INCOMING: ?xml version='1.0' encoding='UTF-8'?stream:stream from=openfire.jocan.local id=3oygw xmlns=jabber:component:accept xmlns:stream=http://etherx.jabber.org/streams; version=1.0stream:error xmlns:stream=http://etherx.jabber.org/streams;bad-namespace-prefix xmlns=urn:ietf:params:xml:ns:xmpp-streams//stream:error [Jul 6 15:15:40] JABBER: asterisk OUTGOING: handshakecccff622b0bafbf9db1e22034292e62610d93f48/handshake [Jul 6 15:15:40] JABBER: asterisk INCOMING: /stream:stream I don't know why connecting my Asterisk to my OpenFire (192.168.2.5) fails... Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem configuring TDM400
Mike wrote: Folks, I have a Xen Asterisk VM with a TDM400 card. When I try to run dahdi_cfg, I get: tempest:~# dahdi_cfg -vvv DAHDI Tools Version - 2.2.0 DAHDI Version: 2.2.0 Echo Canceller(s): Configuration == Channel map: Channel 01: FXO Kewlstart (Default) (Echo Canceler: none) (Slaves: 01) Channel 03: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 03) Channel 04: FXS Kewlstart (Default) (Echo Canceler: none) (Slaves: 04) 3 channels to configure. DAHDI_CHANCONFIG failed on channel 1: No such device or address (6) The card appears to be detected: tempest:~# lspci 00:00.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface I have the kernel modules loaded: tempest:~# lsmod Module Size Used by wctdm 35024 0 dahdi 185352 1 wctdm crc_ccitt 2848 1 dahdi autofs418500 0 ipv6 236612 10 ext3 106664 1 jbd43092 1 ext3 mbcache 8260 1 ext3 dm_mirror 16288 0 dm_log 9444 1 dm_mirror dm_snapshot15108 0 dm_mod 47304 3 dm_mirror,dm_log,dm_snapshot raid1 19200 0 md_mod 69180 1 raid1 thermal_sys11624 0 [ 1327.030178] dahdi: Telephony Interface Registered on major 196 [ 1327.030253] dahdi: Version: 2.2.0 I have Googled for this problem and found a lot of people reporting the issue but nobody really having much of an answer! I've seen the issue a few times. The strange thing is that I did have things working but then I rebotoed the box and it seems to have given up. I have a fairly straight forward DAHDI config file which has served me perfectly well in the past. tempest:~# cat /etc/dahdi/system.conf # Autogenerated by /usr/sbin/dahdi_genconf on Fri Jul 3 09:56:12 2009 # If you edit this file and execute /usr/sbin/dahdi_genconf again, # your manual changes will be LOST. # Dahdi Configuration File # # This file is parsed by the Dahdi Configurator, dahdi_cfg # # Global data loadzone= uk defaultzone = uk fxoks=1 fxsks=3,4 I have tried only bringing up certain channels but that still fails. Does anyone have any idea what could be wrong? Mike. Did do all the device files show up in /dev/dahdi/ ? You should have something close to this: r...@jaguar:~# ls -l /dev/dahdi/ total 0 crw-rw 1 asterisk asterisk 196, 1 2009-07-05 09:32 1 crw-rw 1 asterisk asterisk 196, 2 2009-07-05 09:32 2 crw-rw 1 asterisk asterisk 196, 3 2009-07-05 09:32 3 crw-rw 1 asterisk asterisk 196, 4 2009-07-05 09:32 4 crw-rw 1 asterisk asterisk 196, 254 2009-07-05 09:32 channel crw-rw 1 asterisk asterisk 196, 0 2009-07-05 09:32 ctl crw-rw 1 asterisk asterisk 196, 255 2009-07-05 09:32 pseudo crw-rw 1 asterisk asterisk 196, 253 2009-07-05 09:32 timer -Dave ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] false answer on zaptel
Hi, I have an x100p zaptel card with asterisk 1.4. I'm using the system for outgoing calls. My problem is that Answer() is falsely returning while the call is still ringing and was not really answered yet. I've been digging google, wikis but have not found what might be causing this. SIP works fine, this problem seems to be only zaptel specific. I could use the NVLineDetect application but I think this would be a hack around the problem. Before I start fixing the nvlinedetect code so that it compiles and works with asterisk 1.4 I thought I should ask here first. Any suggestions? Thanks, Botond ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] migrate from zaptel to dahdi
Over the weekend I tried to migrate a system from asterisk 1.4.25, libpri 1.4.1 ZAPTEL 1.4.12.1 to asterisk 1.4.25, libpri 1.4.7 (not 1.4.1) and DAHDI 2.2.0 I removed all old zaptel by: mv /etc/zaptel.conf /tmp mv /etc/asterisk/zapata.conf /tmp rm /etc/init.d/zaptel rm /etc/sysconfig/zaptel rm /etc/modprobe.d/zaptel 2 /dev/null /dev/null rm /etc/udev/rules.d/zaptel.rules rm /etc/rc.d/rc*/*zaptel rm /sbin/zt* rm -rf /usr/share/zaptel rm -rf /usr/include/zaptel Then I just did a CLEAN install of dahdi, libpri and asterisk again. After upgrading incoming calls seemed to work just fine. Outgoing calls gave me an error 99 I have a TE205P installed. I did change the extensions.conf to use DAHDI and not Zap. I had to quickly change back as it is a production system. Any thoughts on what might have happened here? I didnt know if have two libpri versions confused things or what? ANy thoughts for the next time I try are appreciated. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP IP-Trunk to be authenticated based on username and password, not IP address
And these mistery appear with Asterisk to Asterisk and does not appear between
Asterisk to other products or from any IP Phone to Asterisk? How?
Just because call came from Asterisk and was sent to Asterisk it is going to
suffer this? While if it was originated from IP Phone then no problem? And if
it is originated from Asterisk and sent for other softswitch product then it is
working without need for registeration and without need to set the IP address
(just it recognize the username and password)!!!
Why the Asterisk Box B does not recognize the username and password of the SIP
call coming from Asterisk Box A, while it can recognize this if the originator
was SIP IP Phone :) ?
Regards
Bilal
--- On Mon, 7/6/09, Thierry Wehr t.w...@widevoip.com wrote:
From: Thierry Wehr t.w...@widevoip.com
Subject: RE: SIP IP-Trunk to be authenticated based on username and password,
not IP address
To: 'bilal ghayyad' bilmar...@yahoo.com
Date: Monday, July 6, 2009, 7:36 AM
These are mistery of Asterisk like
order codecs are negotiated
Thierry Wehr
Projets Spéciaux
t.w...@widevoip.com
Tél: +33 (0)390 400 675
Fax: +33 (0)390 400 676
http://www.widevoip.com
-Message d'origine-
De : bilal ghayyad [mailto:bilmar...@yahoo.com]
Envoyé : lundi 6 juillet 2009 12:56
À : t.w...@widevoip.com
Objet : RE: SIP IP-Trunk to be authenticated based on
username and password,
not IP address
Thanks a lot for your kindly answer and help.
That is fine, but need to register A on B.
The idea that we were able to place calls from Asterisk A
to a softswitch
using SIP trunk without registeration and it worked. But
the softswitch was
not asterisk. So this is possible in the softswitch, and I
would to do same
with the Asterisk.
From the other side, I am surprised about:
Why the SIP IP Phone (like Polycom) can place a call via
Asterisk without
registeration and without setting the IP address in the
host (actually the
host=dynamic), so why this is not possible when Asterisk A
send for Asterisk
B? Why does not to be considered same as SIP IP Phone is
sending for
Asterisk the call and it is not registered on Asterisk (and
its IP is not
set also in the host parameter), but it succeed by the
username and secret
authentication.
Can u help? And thanks a lot for your already helped :)
Regards
Bilal
--- On Mon, 7/6/09, Thierry Wehr t.w...@widevoip.com
wrote:
From: Thierry Wehr t.w...@widevoip.com
Subject: RE: SIP IP-Trunk to be authenticated based on
username and
password, not IP address
To: 'bilal ghayyad' bilmar...@yahoo.com
Date: Monday, July 6, 2009, 6:28 AM
You MUST register one asterisk on the
other one
See examples for config
Asterisk A ( 10.1.1.1 )
Register = interco:passw...@10.1.1.2
[interco]
Type=friend
Username=interco
Secret=password
Host=10.1.1.2
Contect=incoming-from-asterisk_B
Asterisk B ( 10.1.1.2 )
[interco]
Type=friend
Username=interco
Secret=password
Host=dynamic
Contect=incoming-from-asterisk_A
To dial from A to B
Dial(SIP/interco/${EXTEN})
To dial from B to A
Dial(SIP/interco/${EXTEN})
This must work has it is in production on our side
Thierry Wehr
Projets Spéciaux
t.w...@widevoip.com
Tél: +33 (0)390 400 675
Fax: +33 (0)390 400 676
http://www.widevoip.com
-Message d'origine-
De : bilal ghayyad [mailto:bilmar...@yahoo.com]
Envoyé : lundi 6 juillet 2009 10:03
À : t.w...@widevoip.com
Objet : RE: SIP IP-Trunk to be authenticated based
on
username and password,
not IP address
The [] same as username, but Asterisk B reject calls
came
from Asterisk A.
Anything need to be placed in the Dial command?
Why Asterisk B is not able to authenticate the call
came
from Asterisk A
based on the sip username and secret?
Regards
Bilal
--- On Mon, 7/6/09, Thierry Wehr t.w...@widevoip.com
wrote:
From: Thierry Wehr t.w...@widevoip.com
Subject: RE: SIP IP-Trunk to be authenticated
based on
username and
password, not IP address
To: 'bilal ghayyad' bilmar...@yahoo.com
Date: Monday, July 6, 2009, 3:45 AM
Authentication is based on [],
username, fromuser, secret
If [] different from username you must set
fromuser
Thierry Wehr
Projets Spéciaux
t.w...@widevoip.com
Tél: +33 (0)390 400 675
Fax: +33 (0)390 400 676
http://www.widevoip.com
-Message d'origine-
De : bilal ghayyad [mailto:bilmar...@yahoo.com]
Envoyé : lundi 6 juillet 2009 09:10
À : t.w...@widevoip.com
Objet : RE: SIP IP-Trunk to be authenticated
based
on
username and password,
not IP address
That is correct if u mean at destination
Asterisk, but
what
about source
Asterisk? Sure there is something else need to
be
configured in the SIP
Trunk and maybe in the Dial command?
I was think in the fromuser
Re: [asterisk-users] migrate from zaptel to dahdi
On Jul 6, 2009, at 10:00 AM, Jerry Geis wrote: Over the weekend I tried to migrate a system from asterisk 1.4.25, libpri 1.4.1 ZAPTEL 1.4.12.1 to asterisk 1.4.25, libpri 1.4.7 (not 1.4.1) and DAHDI 2.2.0 I removed all old zaptel by: mv /etc/zaptel.conf /tmp mv /etc/asterisk/zapata.conf /tmp rm /etc/init.d/zaptel rm /etc/sysconfig/zaptel rm /etc/modprobe.d/zaptel 2 /dev/null /dev/null rm /etc/udev/rules.d/zaptel.rules rm /etc/rc.d/rc*/*zaptel rm /sbin/zt* rm -rf /usr/share/zaptel rm -rf /usr/include/zaptel Then I just did a CLEAN install of dahdi, libpri and asterisk again. After upgrading incoming calls seemed to work just fine. Outgoing calls gave me an error 99 I have a TE205P installed. I did change the extensions.conf to use DAHDI and not Zap. I had to quickly change back as it is a production system. Any thoughts on what might have happened here? I didnt know if have two libpri versions confused things or what? ANy thoughts for the next time I try are appreciated. Jerry, Check the dahdichanname setting in asterisk.conf. I had the same issue myself - Niles ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] migrate from zaptel to dahdi
Having gone through similar pains myself, I highly recommend going the the SVN asterisk 1.4 branch. I have had far fewer headaches following this path. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Niles Ingalls Sent: Monday, July 06, 2009 9:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] migrate from zaptel to dahdi On Jul 6, 2009, at 10:00 AM, Jerry Geis wrote: Over the weekend I tried to migrate a system from asterisk 1.4.25, libpri 1.4.1 ZAPTEL 1.4.12.1 to asterisk 1.4.25, libpri 1.4.7 (not 1.4.1) and DAHDI 2.2.0 I removed all old zaptel by: mv /etc/zaptel.conf /tmp mv /etc/asterisk/zapata.conf /tmp rm /etc/init.d/zaptel rm /etc/sysconfig/zaptel rm /etc/modprobe.d/zaptel 2 /dev/null /dev/null rm /etc/udev/rules.d/zaptel.rules rm /etc/rc.d/rc*/*zaptel rm /sbin/zt* rm -rf /usr/share/zaptel rm -rf /usr/include/zaptel Then I just did a CLEAN install of dahdi, libpri and asterisk again. After upgrading incoming calls seemed to work just fine. Outgoing calls gave me an error 99 I have a TE205P installed. I did change the extensions.conf to use DAHDI and not Zap. I had to quickly change back as it is a production system. Any thoughts on what might have happened here? I didnt know if have two libpri versions confused things or what? ANy thoughts for the next time I try are appreciated. Jerry, Check the dahdichanname setting in asterisk.conf. I had the same issue myself - Niles ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] migrate from zaptel to dahdi
Jerry, Check the dahdichanname setting in asterisk.conf. I had the same issue myself - Niles Niles, I did a grep -i dahdichanname /etc/asterisk/* and no results. Jerry ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF is not working occasionally over IAX Trunk
Hi all, Did some more digging in. I changed the trunk from IAX to SIP and still there are not much difference. So I guess it's not an IAX problem. I have enabled DTMF logging and captured the DTMF logs for two servers. (A: where E1 card is connected asterisk-1.4.25, dahdi-linux-2.1.0.4) and B (v1.6.0.9) where IVR is running. I have just pressed * 3 3 but to my untrained eyes it seems asterisk is seeing * * 3 3 3 logs in A: [ TYPE: Control (4) SUBCLASS: Answer (4) ] [SIP/a16-in1-081af4c8] [ TYPE: DTMF Begin (12) SUBCLASS: * (42) ] [DAHDI/37-1] [ TYPE: DTMF Begin (12) SUBCLASS: * (42) ] [SIP/a16-in1-081af4c8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: * (42) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: * (42) ] [SIP/a16-in1-081af4c8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-081af4c8] [ TYPE: DTMF Begin (12) SUBCLASS: 3 (51) ] [DAHDI/37-1] [ TYPE: DTMF Begin (12) SUBCLASS: 3 (51) ] [SIP/a16-in1-081af4c8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: 3 (51) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: 3 (51) ] [SIP/a16-in1-081af4c8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-081af4c8] [ TYPE: DTMF Begin (12) SUBCLASS: 3 (51) ] [DAHDI/37-1] [ TYPE: DTMF Begin (12) SUBCLASS: 3 (51) ] [SIP/a16-in1-081af4c8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: 3 (51) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: 3 (51) ] [SIP/a16-in1-081af4c8] [ TYPE: DTMF Begin (12) SUBCLASS: 3 (51) ] [DAHDI/37-1] [ TYPE: DTMF Begin (12) SUBCLASS: 3 (51) ] [SIP/a16-in1-081af4c8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: 3 (51) ] [DAHDI/37-1] [ TYPE: DTMF End (1) SUBCLASS: 3 (51) ] [SIP/a16-in1-081af4c8] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-081af4c8] logs in B: Over the SIP channel it seems B is getting * 3 3 3 [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: DTMF End (1) SUBCLASS: * (42) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: DTMF End (1) SUBCLASS: 3 (51) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [SIP/a16-in1-09815028] [ TYPE: Null Frame (5)
Re: [asterisk-users] Asterisk Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error
Try client instead of component. Make sure that you selected the component in the menu select as well I can assure you that it works, and that it works well. We use it ;) Julian jonas kellens wrote: I have installed gnutls and gnutls-devel from RedHat repositories [r...@asterisk asterisk]# yum install gnutls gnutls-devel I have installed iksemel with gnutls support : [r...@asterisk asterisk]# cd /usr/src/iksemel-1.3/ [r...@asterisk asterisk]# ./configure --with-gnutls --prefix=/usr [r...@asterisk asterisk]# make [r...@asterisk asterisk]# make check [r...@asterisk asterisk]# make install [r...@asterisk asterisk]# ls -l /usr/lib | grep iksemel -rw-r--r-- 1 root root 184210 2009-07-06 14:52 libiksemel.a -rwxr-xr-x 1 root root 816 2009-07-06 14:52 libiksemel.la lrwxrwxrwx 1 root root 19 2009-07-06 14:52 libiksemel.so - libiksemel.so.3.1.0 lrwxrwxrwx 1 root root 19 2009-07-06 14:52 libiksemel.so.3 - libiksemel.so.3.1.0 -rwxr-xr-x 1 root root 138938 2009-07-06 14:52 libiksemel.so.3.1.0 Then compiled Asterisk again : [r...@asterisk asterisk]# cd /usr/src/asterisk-1.4.25.1/ [r...@asterisk asterisk]# make clean [r...@asterisk asterisk]# ./configure [r...@asterisk asterisk]# make menuconfig [r...@asterisk asterisk]# make [r...@asterisk asterisk]# make install Then edited jabber.conf : [general] debug=yes ;;Turn on debugging by default. autoprune=no;;Auto remove users from buddy list. autoregister=yes;;Auto register users from buddy list. [asterisk] ;;label type=component ;;Client or Component connection serverhost=192.168.2.5 ;;Route to server for example talk.google.com username=aster...@192.168.2.5 ;;Username with optional roster. secret=XX ;;Password port=5222 ;;Port to use defaults to 5222 usetls=yes ;;Use tls or not ;usesasl=yes;;Use sasl or not statusmessage=I am Asterisk ;;Have custom status message for Asterisk. ;timeout=100;;Timeout on the message stack. Then start Asterisk : [r...@asterisk asterisk]# /usr/sbin/asterisk -c And this is the error concerning jabber when wanting to connect to my OpenFire-server: [Jul 6 15:15:36] JABBER: reconnecting. [Jul 6 15:15:36] JABBER: asterisk OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:component:accept' to='aster...@192.168.2.5' version='1.0' [Jul 6 15:15:36] JABBER: asterisk INCOMING: ?xml version='1.0' encoding='UTF-8'?stream:stream from=openfire.jocan.local id=7pI2f xmlns=jabber:component:accept xmlns:stream=http://etherx.jabber.org/streams; version=1.0stream:error xmlns:stream=http://etherx.jabber.org/streams;bad-namespace-prefix xmlns=urn:ietf:params:xml:ns:xmpp-streams//stream:error [Jul 6 15:15:36] JABBER: asterisk OUTGOING: handshake2313234e99edf2891db7901990cf854e8e5639c3/handshake [Jul 6 15:15:36] JABBER: asterisk INCOMING: /stream:stream [Jul 6 15:15:40] WARNING[23732]: res_jabber.c:1573 aji_recv_loop: JABBER: socket read error [Jul 6 15:15:40] JABBER: reconnecting. [Jul 6 15:15:40] JABBER: asterisk OUTGOING: ?xml version='1.0'?stream:stream xmlns:stream='http://etherx.jabber.org/streams' xmlns='jabber:component:accept' to='aster...@192.168.2.5' version='1.0' [Jul 6 15:15:40] JABBER: asterisk INCOMING: ?xml version='1.0' encoding='UTF-8'?stream:stream from=openfire.jocan.local id=3oygw xmlns=jabber:component:accept xmlns:stream=http://etherx.jabber.org/streams; version=1.0stream:error xmlns:stream=http://etherx.jabber.org/streams;bad-namespace-prefix xmlns=urn:ietf:params:xml:ns:xmpp-streams//stream:error [Jul 6 15:15:40] JABBER: asterisk OUTGOING: handshakecccff622b0bafbf9db1e22034292e62610d93f48/handshake [Jul 6 15:15:40] JABBER: asterisk INCOMING: /stream:stream I don't know why connecting my Asterisk to my OpenFire (192.168.2.5) fails... Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Small site survivability
We are currently moving away from a wide-spread Cisco CallManager deployment to Asterisk. For many of our small sites we have the routers configured for what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP registrar. We are converting to SIP, and from what I can tell Cisco wants a license for each router to run SRST over SIP... So my question to the group is: What are you doing for survivability in these small (6-30 phone) sites? I would like to avoid deploying a lot of servers if at all possible. The requirements would be a simple, easy to manage device for the phones to register to in case of WAN failure with 1 or 2 POTS lines attached (also used for 911 calls from that site). Thanks for any suggestions! -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold
Julien Claassen wrote: Hello! I've configured Music on Hold in asterisk, the only, most certainly, stupid problem I have is, which DTMFs to send to activate and deactivate it. If I use the cli, I can establish a call with originate. With the misdn send digit command I can send a number of digits to the other party. But what are the combinations to put the other one on hold? Or do I have to use a completely different mechanism? Any help here is appreciated. A pointer to the right part of the documentation is completely sufficient. Warm regards Julien Putting a person on hold using DTMF is part of the feature code mechanism. You configure it in features.conf. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] false answer on zaptel
Botond Botyanszki wrote: Hi, I have an x100p zaptel card with asterisk 1.4. I'm using the system for outgoing calls. My problem is that Answer() is falsely returning while the call is still ringing and was not really answered yet. I've been digging google, wikis but have not found what might be causing this. SIP works fine, this problem seems to be only zaptel specific. I could use the NVLineDetect application but I think this would be a hack around the problem. Before I start fixing the nvlinedetect code so that it compiles and works with asterisk 1.4 I thought I should ask here first. Any suggestions? Thanks, Botond What Telco are you using? Do you have callprogress=yes or hanguponpolarityswitch=yes in your zapata/dahdi .conf? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial cmd help
On Monday 06 July 2009 12:15:03 am Joseph L. Casale wrote:
exten = s,n,ExecIf($[${ARG1} = 1${ARG1:1}
]?Set(Dialnum=${ARG1:1}):Set(Dialnum=${ARG1}))
Much simpler Dhaval, thanks!
Even simpler:
exten = s,n,Set(Dialnum=${IF($[${ARG1:0:1}=1]?${ARG1:1}:${ARG1})})
--
Tilghman
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Re: [asterisk-users] Small site survivability
Audiocodes supports SRST on their mediapack analog gateways. Cory J. Andrews Director New Market Initiatives Sayers Media Group VoIP Supply, LLC 454 Sonwil Drive Buffalo, NY 14225 716-250-3402 OFFICE 716-630-1548 FAX 716-601-4474 MOBILE candr...@sayersmedia.com mailto:br...@voipsupply.com Have I exceeded your expectations? Please share your experience with my boss, Benjamin P. Sayers mailto:bsay...@voipsupply.com , CEO NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonathan Thurman Sent: Monday, July 06, 2009 11:28 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Small site survivability We are currently moving away from a wide-spread Cisco CallManager deployment to Asterisk. For many of our small sites we have the routers configured for what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP registrar. We are converting to SIP, and from what I can tell Cisco wants a license for each router to run SRST over SIP... So my question to the group is: What are you doing for survivability in these small (6-30 phone) sites? I would like to avoid deploying a lot of servers if at all possible. The requirements would be a simple, easy to manage device for the phones to register to in case of WAN failure with 1 or 2 POTS lines attached (also used for 911 calls from that site). Thanks for any suggestions! -Jonathan ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to notify that a message is waiting using RingTones with a Dahdi-connected analog phone ?
On Monday 06 July 2009 08:18:37 am Dave Fullerton wrote: Olivier wrote: Hi, I'm wondering how I could notify to a dumb analog phone that a voicemail message is waiting. My goal would be to change the tone that is heard just before user starts to dial. Any idea on that ? Yea, it's called stutter dial tone. For DAHDI channels just specify the mailbox in chan_dahdi.conf. If it's connected to an ATA then specify the mailbox on the peer in sip.conf/iax.conf. Additionally, if the OP wanted to change the default tones, those are specified in indications.conf. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Variable using AMI
Hello, if I do a variable assignation using AMI interface, that variable will be visible only for the current AMI instance or will be readable for all AMI instances?. I will login using the same user, concurrently. A program will write a global variable using the same name and if asterisk don't have any scope rules I have to find another way to do what I want. Thanks in advance. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_mobile help.
Thanks for your response. I gave loads of info in my original mail, surely someone can help without jumping distro? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] dahdi_dummy configure
In article 20090703192211.gq25...@xorcom.com, Tzafrir Cohen tzafrir.co...@xorcom.com wrote: On Fri, Jul 03, 2009 at 02:38:46PM -0400, Jerry Geis wrote: Description Alarms IRQ bpviol CRC4 DAHDI_DUMMY/1 (source: Linux26) 1UNCONFIGUR 0 0 0 Is there a way to configure dahdi_dummy so that status reports OK instead of unconfigured. There is no need to configure dahdi_dummy. Perhaps it should report OK then, instead of UNCONFIGURED Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_mobile help.
Try upgrading your bluez library. You can also try a parallel installation with the last revision of chan_mobile. Use the same phone always to discard any phone issues. On Mon, Jul 6, 2009 at 11:43 AM, Razza razz...@gmail.com wrote: Thanks for your response. I gave loads of info in my original mail, surely someone can help without jumping distro? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk capacity
Un-top-posting... On Mon, 6 Jul 2009, abdelkader wrote: I need to know how many calls I can handle with my Asterisk. Steve Totaro schrieb: It can make 9977.39 Bogocalls of course! On Mon, 6 Jul 2009, Philipp Kempgen wrote: Mind to share the formula? Wait. Got it. Bogomips/2. Why on earth isn't that documented?! ;) Because your formula is incomplete. Bogocalls are a meaningless unit of measurement frequently applied to imponderable questions posed by people who lack sufficient knowledge to ask questions that can yield meaningful answers. While a useful unit of measure in an in-Prefect world, you have to apply the per second conversion factor by dividing the number of bogocalls by 237.5569 to get the number of real-world calls per second. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Source for OpenVox cards?
In article be95b6c50907050425h777a9eha25924d88b5ba...@mail.gmail.com, Timothy Legge timle...@gmail.com wrote: I am looking for a source for an OpenVox card. Has anyone purchased through http://www.voiplink.com or do you normally use another vendor or OpenVox.cn directly? thanks Tim I have used voipon.co.uk, but I don't know whether that's useful to you, as you didn't say which country you are in. Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Small site survivability
On Mon, 6 Jul 2009, Jonathan Thurman wrote: We are currently moving away from a wide-spread Cisco CallManager deployment to Asterisk. For many of our small sites we have the routers configured for what Cisco calls SRST so if we have a WAN failure, the router acts as a SCCP registrar. We are converting to SIP, and from what I can tell Cisco wants a license for each router to run SRST over SIP... So my question to the group is: What are you doing for survivability in these small (6-30 phone) sites? I would like to avoid deploying a lot of servers if at all possible. The requirements would be a simple, easy to manage device for the phones to register to in case of WAN failure with 1 or 2 POTS lines attached (also used for 911 calls from that site). Thanks for any suggestions! Deploy a lot of small asterisk based appliances... This way you can completely decentralise your setup and give each office it's own autonomous system, only needing the WAN links for inter-site calls (and maybe your backhaul to the PSTN) 30 phones will trivially work from a diskless, fanless, processor, so no need for anything too clever. If building them yourselves the cost per site ought to be under $600 for the hardware (Under £400 where I am, so apply conversion rate) you could use one of the pre-built packages for this - pbxinaflash, or buy something like an Atcom unit, etc. Gordon___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Listed agents in queue not ringing
Hi All,
I am having a problem when we call inbound the ivr picks up send caller to
the queue but does not forward the calls to the listed agents, however if we
use the call groups instead of queues it rings to the listed agents in group
Here are the default settings
include=DID_suhaib_timeinterval_inbound
include=DID_suhaib_timeinterval_AllTheTime
include=DID_suhaib_timeinterval_OfficeHours
include=DID_suhaib_timeinterval_AfterHours
include=DID_suhaib_timeinterval_Weekend
include=DID_suhaib_timeinterval_EarlyHours
exten=0,1,
exten=o,1,Goto(default,6043,1)
exten=6800,1,VoiceMailMain(${CALLERID(num)}...@default)
Here is the office hour interval settings
exten=__X.,1,Goto(voicemenu-custom-2|s|1)
exten=_XX,1,Goto(queues|6525|1)
Here are the queues settings
exten=6500,1,Queue(${EXTEN})
exten=6501,1,Queue(${EXTEN})
exten=6502,1,Queue(${EXTEN})
exten=6503,1,Queue(${EXTEN})
exten=6509,1,Queue(${EXTEN})
exten=6900,1,agentlogin()
exten=6950,1,agentcallbacklogin()
Here is the queue settings
fullname=TechPC
strategy=ringall
timeout=180
wrapuptime=15
autofill=yes
autopause=no
joinempty=yes
leavewhenempty=no
reportholdtime=yes
maxlen=0
musicclass=default
member=Agent/6029
member=Agent/6038
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Re: [asterisk-users] Variable using AMI
On Mon, Jul 6, 2009 at 11:37 AM, Carlos Ruiz Diazcarlos.ruizd...@gmail.com wrote: Hello, if I do a variable assignation using AMI interface, that variable will be visible only for the current AMI instance or will be readable for all AMI instances?. I will login using the same user, concurrently. A program will write a global variable using the same name and if asterisk don't have any scope rules I have to find another way to do what I want. If you want to maintain scope for a variable across multiple calls you should maintain the value of that variable outside of asterisk and keep setting it for each new phonecall. Global variables in asterisk do not do what you are describing. AMI does have something where you can name a particular AMI session, and then communication for that session will care that name. That should not be confused with a system-wide global variable. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold
Thanks Brent! I'll have a look there in features.conf. Warm regards Julien Music was my first love and it will be my last (John Miles) FIND MY WEB-PROJECT AT: http://ltsb.sourceforge.net the Linux TextBased Studio guide === AND MY PERSONAL PAGES AT: === http://www.juliencoder.de ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] chan_mobile help.
I'm running centos, so tried a yum upgrade but nothing was marked for upgrade. I've reinstalled bluez-libs.i386 0:3.7-1.1. I've tried a different dongle, but still get the same message. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] migrate from zaptel to dahdi
On Mon, Jul 06, 2009 at 10:00:12AM -0400, Jerry Geis wrote: After upgrading incoming calls seemed to work just fine. Outgoing calls gave me an error 99 What is the outout of: cat /proc/dahdi/* Can you provide a trace of such a failed call? -- Tzafrir Cohen icq#16849755 jabber:tzafrir.co...@xorcom.com +972-50-7952406 mailto:tzafrir.co...@xorcom.com http://www.xorcom.com iax:gu...@local.xorcom.com/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error
On Mon, 2009-07-06 at 16:18 +0100, Julian Lyndon-Smith wrote: I can assure you that it works, and that it works well. We use it ;) My jabber.conf : [general] debug=yes ;;Turn on debugging by default. autoprune=no;;Auto remove users from buddy list. autoregister=yes;;Auto register users from buddy list. [asterisk] ;;label type=client ;;Client or Component connection serverhost=192.168.2.5 ;;Route to server for example talk.google.com username=aster...@192.168.2.5 ;;Username with optional roster. secret=XX ;;Password port=5222 ;;Port to use defaults to 5222 usetls=yes ;;Use tls or not usesasl=yes ;;Use sasl or not statusmessage=I am Asterisk ;;Have custom status message for Asterisk. ;timeout=100;;Timeout on the message stack. Then I get the following : [Jul 6 20:07:57] JABBER: asterisk INCOMING: ?xml version='1.0' encoding='UTF-8'?stream:stream xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:client from=openfire.jocan.local id=56ff9859 xml:lang=en version=1.0stream:featuresmechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismDIGEST-MD5/mechanismmechanismPLAIN/mechanismmechanismANONYMOUS/mechanismmechanismCRAM-MD5/mechanism/mechanismscompression xmlns=http://jabber.org/features/compress;methodzlib/method/compressionauth xmlns=http://jabber.org/features/iq-auth/register xmlns=http://jabber.org/features/iq-register//stream:features [Jul 6 20:07:57] JABBER: asterisk OUTGOING: auth xmlns='urn:ietf:params:xml:ns:xmpp-sasl' mechanism='DIGEST-MD5'/ [Jul 6 20:07:57] JABBER: asterisk INCOMING: challenge xmlns=urn:ietf:params:xml:ns:xmpp-saslcmVhbG09Im9wZW5maXJlLmpvY2FuLmxvY2FsIixub25jZT0iSngyRVZCRmlDNlI4K1hlMU5rbm9PUUNWT1VEN1pGMEpXcnRydUxjdiIscW9wPSJhdXRoIixjaGFyc2V0PXV0Zi04LGFsZ29yaXRobT1tZDUtc2Vzcw==/challenge [Jul 6 20:07:57] JABBER: asterisk OUTGOING: response xmlns='urn:ietf:params:xml:ns:xmpp-sasl'dXNlcm5hbWU9ImFzdGVyaXNrIixyZWFsbT0ib3BlbmZpcmUuam9jYW4ubG9jYWwiLG5vbmNlPSJKeDJFVkJGaUM2UjgrWGUxTmtub09RQ1ZPVUQ3WkYwSldydHJ1TGN2Iixjbm9uY2U9IjQzZTVmYjFkNjZiMTU2OGI1MDFjNzk0ZDQ0MzMyYzFiIixuYz0wMDAwMDAwMSxxb3A9YXV0aCxkaWdlc3QtdXJpPSJ4bXBwLzE5Mi4xNjguMi41IixyZXNwb25zZT1kNGUxYzQ0ZDM0OGNjNWJkN2E2MzJiNzdmZjRjZTQ0OCxjaGFyc2V0PXV0Zi04/response [Jul 6 20:07:57] JABBER: asterisk INCOMING: failure xmlns=urn:ietf:params:xml:ns:xmpp-saslnot-authorized//failure [Jul 6 20:07:57] ERROR[24565]: res_jabber.c:606 aji_act_hook: JABBER: encryption failure. possible bad password. I am 100% sure I have the correct password ! I even took a very simple password without any special characters... Can you advise ?? Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable using AMI
Maybe you could use the Asterisk Database. In 1.4 you can do it with DBGet and DBPut: http://www.voip-info.org/wiki/view/Asterisk+cmd+DBget http://www.voip-info.org/wiki/view/Asterisk+cmd+DBput In 1.6 use DB() function. Regards, Juan David Backeberg wrote: On Mon, Jul 6, 2009 at 11:37 AM, Carlos Ruiz Diazcarlos.ruizd...@gmail.com wrote: Hello, if I do a variable assignation using AMI interface, that variable will be visible only for the current AMI instance or will be readable for all AMI instances?. I will login using the same user, concurrently. A program will write a global variable using the same name and if asterisk don't have any scope rules I have to find another way to do what I want. If you want to maintain scope for a variable across multiple calls you should maintain the value of that variable outside of asterisk and keep setting it for each new phonecall. Global variables in asterisk do not do what you are describing. AMI does have something where you can name a particular AMI session, and then communication for that session will care that name. That should not be confused with a system-wide global variable. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial cmd help
Even simpler:
exten = s,n,Set(Dialnum=${IF($[${ARG1:0:1}=1]?${ARG1:1}:${ARG1})})
Thanks Tilghman,
I am making a note of this as well!
jlc
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Re: [asterisk-users] Asterisk Jabber : WARNING: res_jabber.c aji_recv_loop: JABBER: socket read error
usetls=no Julian jonas kellens wrote: On Mon, 2009-07-06 at 16:18 +0100, Julian Lyndon-Smith wrote: I can assure you that it works, and that it works well. We use it ;) My jabber.conf : [general] debug=yes ;;Turn on debugging by default. autoprune=no;;Auto remove users from buddy list. autoregister=yes;;Auto register users from buddy list. [asterisk] ;;label type=client ;;Client or Component connection serverhost=192.168.2.5 ;;Route to server for example talk.google.com username=aster...@192.168.2.5 ;;Username with optional roster. secret=XX ;;Password port=5222 ;;Port to use defaults to 5222 usetls=yes ;;Use tls or not usesasl=yes ;;Use sasl or not statusmessage=I am Asterisk ;;Have custom status message for Asterisk. ;timeout=100;;Timeout on the message stack. Then I get the following : [Jul 6 20:07:57] JABBER: asterisk INCOMING: ?xml version='1.0' encoding='UTF-8'?stream:stream xmlns:stream=http://etherx.jabber.org/streams; xmlns=jabber:client from=openfire.jocan.local id=56ff9859 xml:lang=en version=1.0stream:featuresmechanisms xmlns=urn:ietf:params:xml:ns:xmpp-saslmechanismDIGEST-MD5/mechanismmechanismPLAIN/mechanismmechanismANONYMOUS/mechanismmechanismCRAM-MD5/mechanism/mechanismscompression xmlns=http://jabber.org/features/compress;methodzlib/method/compressionauth xmlns=http://jabber.org/features/iq-auth/register xmlns=http://jabber.org/features/iq-register//stream:features [Jul 6 20:07:57] JABBER: asterisk OUTGOING: auth xmlns='urn:ietf:params:xml:ns:xmpp-sasl' mechanism='DIGEST-MD5'/ [Jul 6 20:07:57] JABBER: asterisk INCOMING: challenge xmlns=urn:ietf:params:xml:ns:xmpp-saslcmVhbG09Im9wZW5maXJlLmpvY2FuLmxvY2FsIixub25jZT0iSngyRVZCRmlDNlI4K1hlMU5rbm9PUUNWT1VEN1pGMEpXcnRydUxjdiIscW9wPSJhdXRoIixjaGFyc2V0PXV0Zi04LGFsZ29yaXRobT1tZDUtc2Vzcw==/challenge [Jul 6 20:07:57] JABBER: asterisk OUTGOING: response xmlns='urn:ietf:params:xml:ns:xmpp-sasl'dXNlcm5hbWU9ImFzdGVyaXNrIixyZWFsbT0ib3BlbmZpcmUuam9jYW4ubG9jYWwiLG5vbmNlPSJKeDJFVkJGaUM2UjgrWGUxTmtub09RQ1ZPVUQ3WkYwSldydHJ1TGN2Iixjbm9uY2U9IjQzZTVmYjFkNjZiMTU2OGI1MDFjNzk0ZDQ0MzMyYzFiIixuYz0wMDAwMDAwMSxxb3A9YXV0aCxkaWdlc3QtdXJpPSJ4bXBwLzE5Mi4xNjguMi41IixyZXNwb25zZT1kNGUxYzQ0ZDM0OGNjNWJkN2E2MzJiNzdmZjRjZTQ0OCxjaGFyc2V0PXV0Zi04/response [Jul 6 20:07:57] JABBER: asterisk INCOMING: failure xmlns=urn:ietf:params:xml:ns:xmpp-saslnot-authorized//failure [Jul 6 20:07:57] ERROR[24565]: res_jabber.c:606 aji_act_hook: JABBER: encryption failure. possible bad password. I am 100% sure I have the correct password ! I even took a very simple password without any special characters... Can you advise ?? Jonas. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Get channel string
Hello, When I attempt to make a call using AMI interface with originate action I successfully specify all of the needed parameters but when I try to control the flow of the call I am unable to identify each call because asterisk uses some kind of unique identification appended to the channel string. E.g. channel: SIP/1000 results in SIP/1000-*0845ea38*. I also found an auto-generated unique ID but I don't know how to retrieve it immediately after the originate action to be able to use it to identify the calls that I made. How can I get the actual channel string after calling Originate? or how can I get the unique ID of a call about to start (or already started) using the same action (Originate). Regards. Carlos. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable using AMI
Thank you! I did not know the existence of DB command. The command allows me to store KVPs but I have to use the same variable name every time so every process that starts the AMI instance will override the values making it unusable for what I want to achieve. It was really useful anyways. :) 2009/7/6 Juan E. Rodríguez jerdg...@gmail.com Maybe you could use the Asterisk Database. In 1.4 you can do it with DBGet and DBPut: http://www.voip-info.org/wiki/view/Asterisk+cmd+DBget http://www.voip-info.org/wiki/view/Asterisk+cmd+DBput In 1.6 use DB() function. Regards, Juan David Backeberg wrote: On Mon, Jul 6, 2009 at 11:37 AM, Carlos Ruiz Diazcarlos.ruizd...@gmail.com carlos.ruizd...@gmail.com wrote: Hello, if I do a variable assignation using AMI interface, that variable will be visible only for the current AMI instance or will be readable for all AMI instances?. I will login using the same user, concurrently. A program will write a global variable using the same name and if asterisk don't have any scope rules I have to find another way to do what I want. If you want to maintain scope for a variable across multiple calls you should maintain the value of that variable outside of asterisk and keep setting it for each new phonecall. Global variables in asterisk do not do what you are describing. AMI does have something where you can name a particular AMI session, and then communication for that session will care that name. That should not be confused with a system-wide global variable. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get channel string
Carlos Ruiz Diaz schrieb: When I attempt to make a call using AMI interface with originate action I successfully specify all of the needed parameters but when I try to control the flow of the call I am unable to identify each call because asterisk uses some kind of unique identification appended to the channel string. E.g. channel: SIP/1000 results in SIP/1000-*0845ea38*. I also found an auto-generated unique ID but I don't know how to retrieve it immediately after the originate action to be able to use it to identify the calls that I made. How can I get the actual channel string after calling Originate? or how can I get the unique ID of a call about to start (or already started) using the same action (Originate). Doesn't the OriginateResponse give you that information? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get channel string
No :( . Response gave me an empty unique-Id. Apparently it is generated on the fly once the resources are allocated or something else. I don't have any channel information in the response. On Mon, Jul 6, 2009 at 3:22 PM, Philipp Kempgen philipp.kemp...@amooma.dewrote: Carlos Ruiz Diaz schrieb: When I attempt to make a call using AMI interface with originate action I successfully specify all of the needed parameters but when I try to control the flow of the call I am unable to identify each call because asterisk uses some kind of unique identification appended to the channel string. E.g. channel: SIP/1000 results in SIP/1000-*0845ea38*. I also found an auto-generated unique ID but I don't know how to retrieve it immediately after the originate action to be able to use it to identify the calls that I made. How can I get the actual channel string after calling Originate? or how can I get the unique ID of a call about to start (or already started) using the same action (Originate). Doesn't the OriginateResponse give you that information? Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable using AMI
Well, I do not understand very well what you are trying to do, but I'll give you some advice: If you want a variable only for the AGI you call, you just have to declare that variable on the AGI. If you would like to make visible that variable as long as the call is active and for each call, even if the name is the same, you have to set a channel variable with the Set(variable=value) command. If you would like to have a variable shared between two or more channels, use the SHARED() funcion(Asterisk 1.6, back ported to 1.4) If you want a variable to be accessed from all the channels, you could use a global variable. http://www.voip-info.org/wiki/view/Asterisk+variables Regards, Juan Carlos Ruiz Diaz wrote: Thank you! I did not know the existence of DB command. The command allows me to store KVPs but I have to use the same variable name every time so every process that starts the AMI instance will override the values making it unusable for what I want to achieve. It was really useful anyways. :) 2009/7/6 Juan E. Rodríguez jerdg...@gmail.com mailto:jerdg...@gmail.com Maybe you could use the Asterisk Database. In 1.4 you can do it with DBGet and DBPut: http://www.voip-info.org/wiki/view/Asterisk+cmd+DBget http://www.voip-info.org/wiki/view/Asterisk+cmd+DBput In 1.6 use DB() function. Regards, Juan David Backeberg wrote: On Mon, Jul 6, 2009 at 11:37 AM, Carlos Ruiz Diazcarlos.ruizd...@gmail.com mailto:carlos.ruizd...@gmail.com wrote: Hello, if I do a variable assignation using AMI interface, that variable will be visible only for the current AMI instance or will be readable for all AMI instances?. I will login using the same user, concurrently. A program will write a global variable using the same name and if asterisk don't have any scope rules I have to find another way to do what I want. If you want to maintain scope for a variable across multiple calls you should maintain the value of that variable outside of asterisk and keep setting it for each new phonecall. Global variables in asterisk do not do what you are describing. AMI does have something where you can name a particular AMI session, and then communication for that session will care that name. That should not be confused with a system-wide global variable. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Variable using AMI
I am sorry for my bad English. Apparently I'm explain myself wrongly but you got the point. I tried GetVar as AMI action but I have to specify a channel string. Of course I have the channel string, I parametrized it but Asterisk adds another string to the original channel and I can't obtain the variable value because of the lack of the real channel string. eg. I write SIP/1000 in the channel param. but asterisk adds SIP/1000-*12eg12*. Obviously is not always the same string. This is just a illustrative demonstration. I wrote a mail asking for help with that problem. Thanks. 2009/7/6 Juan E. Rodríguez jerdg...@gmail.com Well, I do not understand very well what you are trying to do, but I'll give you some advice: If you want a variable only for the AGI you call, you just have to declare that variable on the AGI. If you would like to make visible that variable as long as the call is active and for each call, even if the name is the same, you have to set a channel variable with the Set(variable=value) command. If you would like to have a variable shared between two or more channels, use the SHARED() funcion(Asterisk 1.6, back ported to 1.4) If you want a variable to be accessed from all the channels, you could use a global variable. http://www.voip-info.org/wiki/view/Asterisk+variables Regards, Juan Carlos Ruiz Diaz wrote: Thank you! I did not know the existence of DB command. The command allows me to store KVPs but I have to use the same variable name every time so every process that starts the AMI instance will override the values making it unusable for what I want to achieve. It was really useful anyways. :) 2009/7/6 Juan E. Rodríguez jerdg...@gmail.com Maybe you could use the Asterisk Database. In 1.4 you can do it with DBGet and DBPut: http://www.voip-info.org/wiki/view/Asterisk+cmd+DBget http://www.voip-info.org/wiki/view/Asterisk+cmd+DBput In 1.6 use DB() function. Regards, Juan David Backeberg wrote: On Mon, Jul 6, 2009 at 11:37 AM, Carlos Ruiz Diazcarlos.ruizd...@gmail.com carlos.ruizd...@gmail.com wrote: Hello, if I do a variable assignation using AMI interface, that variable will be visible only for the current AMI instance or will be readable for all AMI instances?. I will login using the same user, concurrently. A program will write a global variable using the same name and if asterisk don't have any scope rules I have to find another way to do what I want. If you want to maintain scope for a variable across multiple calls you should maintain the value of that variable outside of asterisk and keep setting it for each new phonecall. Global variables in asterisk do not do what you are describing. AMI does have something where you can name a particular AMI session, and then communication for that session will care that name. That should not be confused with a system-wide global variable. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some IAX calls do not disconnect.
Ah, and you are using iax trunking - which depends on the realtime clock. I'm no expert on virtualization, but I think I read that the usb based zaptel clock was a better choice in a virtualized system. T. On 6 Jul 2009, at 06:44, Rajkumar S wrote: Hi, The servers B C are running in a virtual machine (linux kvm) and uses ztdummy for timing. Server A has a digium card. I am not sure if this is the cause of the problems I am facing. raj On Fri, Jul 3, 2009 at 5:57 PM, Rajkumar Srajkum...@gmail.com wrote: On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote: I'd try adding transfer=no in the B iax.conf This does not help, I still have some ghost calls in B a16-in1*CLI core show channels Channel Location State Application(Data) IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-12174 outbo...@inbound-cal Up Dial(iax2/a16-in1- sangoma-flip IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-7161outbo...@inbound-cal Up Dial(iax2/a16-in1- sangoma-flip IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-14813 s...@queue:20 Up Dial(iax2/a16-in1- a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-4485s...@queue:20 Up Dial(iax2/a16-in1- a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-10115 s...@queue:20 Up Dial(iax2/a16-in1- a16-q1/queue 10 active channels 5 active calls raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bug or Not?
Hi gang, When I try to park a call using blind-transfer (#1), the caller hears the lot instead of the transferring party. Attended transfer and blind transfer from the phone buttons (Polycom 501) work fine, so this isn't a showstopper, just a WHY??. Thanks for you attention. Danny Nicholas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] What is the best way to share extension state
http://bugs.digium.com/view.php?id=14595 has a patch to add a new class, bridge, so you get less events in AMI. This is for 1.6.0.x. It will give you an idea of what needs to be changed in order to make the call class of messages more granular. -- Jim Dickenson mailto:dicken...@cfmc.com CfMC http://www.cfmc.com/ From: Philipp Kempgen philipp.kemp...@amooma.de Organization: Amooma GmbH Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Mon, 06 Jul 2009 13:36:52 +0200 To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] What is the best way to share extension state Benny Amorsen schrieb: asterisk-us...@rogg.is writes: 3) Hook into the AMI and parse the ExtensionStatusEvent which I think gives me what I want. The challenge with AMI is that it is becoming a high-bandwidth channel. If you're only interested in one event type, you will spend quite a bit of CPU time just discarding uninteresting events. Perhaps it would be possible to make events more granular, instead of just on/off? After you're logged in send Action: Events EventMask: call or Action: Events EventMask: call,system if you're interested in reload etc. But you are right. call still gives you events like NewExten, NewChannel etc. apart from ExtensionStatus. NewExten can be pretty verbose. Philipp Kempgen -- AMOOMA GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Geschäftsführer: Stefan Wintermeyer, Handelsregister: Neuwied B14998 Asterisk: http://the-asterisk-book.com - http://das-asterisk-buch.de Videos of the AMOOCON VoIP conference 2009 - http://www.amoocon.de -- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail attachments not working
Today I discovered that voicemail attachments are not working on our latest
asterisk server (version 1.4.24.1). I have two other asterisk servers that
I maintain but I didn¹t do the configuration on these so this is my first
time that I have done the voicemail.conf. I get an email but there is no
attachment. Maybe there is something else I need to configure that I don¹t
know about? Here is my actual config, the only difference is I removed all
the mailboxes for the purpose of sharing with the world. However, I have
made sure there are not spaces between fields as I hear that causes
problems.
[general]
format = gsm|wav49|wav
attach = yes
serveremail = asterisk
serveremail = nore...@mustangintl.com
mailcmd = /usr/sbin/sendmail -v -t -f aster...@hisg-it.net
maxlogins = 3
emaildateformat = %A, %B %d, %Y at %r
sendvoicemail = yes ; Allow the user to compose and send a voicemail while
inside
emailsubject = [PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
[zonemessages]
eastern = America/New_York|'vm-received' Q 'digits/at' IMp
central = America/Chicago|'vm-received' Q 'digits/at' IMp
central24 = America/Chicago|'vm-received' q 'digits/at' H N 'hours'
military = Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p'
european = Europe/Copenhagen|'vm-received' a d b 'digits/at' HM
[default]
116 = 1149,employee,emplo...@domain.org
Suggestions?
Thank you everyone in advance.
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Re: [asterisk-users] Some IAX calls do not disconnect.
Just use SIP and solve all your problems. On Mon, Jul 6, 2009 at 5:00 PM, Tim Pantont...@westhawk.co.uk wrote: Ah, and you are using iax trunking - which depends on the realtime clock. I'm no expert on virtualization, but I think I read that the usb based zaptel clock was a better choice in a virtualized system. T. On 6 Jul 2009, at 06:44, Rajkumar S wrote: Hi, The servers B C are running in a virtual machine (linux kvm) and uses ztdummy for timing. Server A has a digium card. I am not sure if this is the cause of the problems I am facing. raj On Fri, Jul 3, 2009 at 5:57 PM, Rajkumar Srajkum...@gmail.com wrote: On Fri, Jul 3, 2009 at 12:36 PM, Tim Pantont...@westhawk.co.uk wrote: I'd try adding transfer=no in the B iax.conf This does not help, I still have some ghost calls in B a16-in1*CLI core show channels Channel Location State Application(Data) IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-12174 outbo...@inbound-cal Up Dial(iax2/a16-in1-sangoma-flip IAX2/a16-in1-sangoma (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-7161 outbo...@inbound-cal Up Dial(iax2/a16-in1-sangoma-flip IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-14813 s...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-4485 �...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue IAX2/a16-in1-a16-q1- (None) Up AppDial((Outgoing Line)) IAX2/a16-in1-10115 s...@queue:20 Up Dial(iax2/a16-in1-a16-q1/queue 10 active channels 5 active calls raj ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bug or Not?
'One touch park' was designed to work around this issue. PaulH Danny Nicholas wrote: Hi gang, When I try to park a call using blind-transfer (#1), the caller hears the lot instead of the transferring party. Attended transfer and blind transfer from the phone buttons (Polycom 501) work fine, so this isn’t a showstopper, just a “WHY??”. Thanks for you attention. Danny Nicholas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some IAX calls do not disconnect.
- Steve Totaro stot...@asteriskhelpdesk.com wrote: Just use SIP and solve all your problems. I seem to be noticing a common element to your posts about IAX. :-) I've been successfully using IAX in a large scale environment with no problems... yet. Can you shed some light on the reasoning behind your obvious dislike of IAX2? It is supposed to be the 'killer' of SIP from a usability standpoint (NAT traversal is quick to my mind...). BUT, is it just not robust enough in your experience? Are there inherent problems with the protocol itself? Is this changing now that IAX2 has it's own RFC? Is it the implementation within Asterisk that is the problem? I'm very interested to to know where your disdain comes from. :-) Thanks Steve! --Tim ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some IAX calls do not disconnect.
On Mon, Jul 6, 2009 at 9:41 PM, Tim Nelsontnel...@rockbochs.com wrote: - Steve Totaro stot...@asteriskhelpdesk.com wrote: Just use SIP and solve all your problems. I seem to be noticing a common element to your posts about IAX. :-) I've been successfully using IAX in a large scale environment with no problems... yet. Can you shed some light on the reasoning behind your obvious dislike of IAX2? It is supposed to be the 'killer' of SIP from a usability standpoint (NAT traversal is quick to my mind...). BUT, is it just not robust enough in your experience? Are there inherent problems with the protocol itself? Is this changing now that IAX2 has it's own RFC? Is it the implementation within Asterisk that is the problem? I'm very interested to to know where your disdain comes from. :-) Thanks Steve! --Tim First define large scale. It certainly means different things to different people. Second, It comes from huge amounts of audio problems over many, many years, and many, many implementations. I actually don't have a disdain for it, it has made me a good deal of money by fixing ITSPs/carrier's audio issues by switching them to SIP and still does so I have a fondness for it. Keep up the sub par protocol, it helps with the balance sheet! Third, it will never kill SIP. First of all, Digium owns the name and we have seen what they are willing to do to attack people for trademark or copyright infringement (think about the Google Adwords debacle and the the Open letter to Digium drafted by Trixter that I am not sure was ever fully addressed by Digium.) It would have to be renamed or something. I think the same thing of DAHDI. They want control over the the names Inter Asterisk Exchange and Digium (whatever the heck the rest of it means.) Second, SIP is the industry standard. Only a couple of goofy phones do IAX2 as far as I know, some crappy handsets I wouldn't even bother testing if offered as a free demo unit. SNOM might now, I am not sure but I think I read interest in it or it was actually accomplished. SNOM is OK but I was never a big fan. When I see it on a Polycom, Cisco, NEC, 3Com, or any other major vendor's phones or platforms, then I may rethink my ideas. If 3Com and Digium are partnered up now, how come the NBX for V3000 doesn't support IAX2? They do have SIP. Second, there are work arounds for just about every downfall of SIP, like NAT traversal and the like. Third, ALL REAL TIME VOICE traffic is on a single port. There is a big issue there, I won't elaborate, but just think about it. SIP is here to stay until some other protocol comes about, but certainly not IAX2. It will be along the evolution of H323 to SIP to X., but not IAX,lol. Do you realize that most providers are dropping IAX2 support, even IAX.cc recommends SIP, gotta wonder why? Maybe it is all good now, but I won't bank my reputation on it. I use what I know works well, period. Even unnamed Digium Employees have poo pooed IAX2, albeit a year or two ago. It looks good on paper, didn't perform well historically, and now just like anything that I have lost trust in, it has to earn my trust back and that is not easy. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Some IAX calls do not disconnect.
On Tue, Jul 7, 2009 at 12:05 AM, Steve Totarostot...@totarotechnologies.com wrote: On Mon, Jul 6, 2009 at 9:41 PM, Tim Nelsontnel...@rockbochs.com wrote: - Steve Totaro stot...@asteriskhelpdesk.com wrote: Just use SIP and solve all your problems. I seem to be noticing a common element to your posts about IAX. :-) I've been successfully using IAX in a large scale environment with no problems... yet. Can you shed some light on the reasoning behind your obvious dislike of IAX2? It is supposed to be the 'killer' of SIP from a usability standpoint (NAT traversal is quick to my mind...). BUT, is it just not robust enough in your experience? Are there inherent problems with the protocol itself? Is this changing now that IAX2 has it's own RFC? Is it the implementation within Asterisk that is the problem? I'm very interested to to know where your disdain comes from. :-) Thanks Steve! --Tim First define large scale. It certainly means different things to different people. Second, It comes from huge amounts of audio problems over many, many years, and many, many implementations. I actually don't have a disdain for it, it has made me a good deal of money by fixing ITSPs/carrier's audio issues by switching them to SIP and still does so I have a fondness for it. Keep up the sub par protocol, it helps with the balance sheet! Third, it will never kill SIP. First of all, Digium owns the name and we have seen what they are willing to do to attack people for trademark or copyright infringement (think about the Google Adwords debacle and the the Open letter to Digium drafted by Trixter that I am not sure was ever fully addressed by Digium.) It would have to be renamed or something. I think the same thing of DAHDI. They want control over the the names Inter Asterisk Exchange and Digium (whatever the heck the rest of it means.) Second, SIP is the industry standard. Only a couple of goofy phones do IAX2 as far as I know, some crappy handsets I wouldn't even bother testing if offered as a free demo unit. SNOM might now, I am not sure but I think I read interest in it or it was actually accomplished. SNOM is OK but I was never a big fan. When I see it on a Polycom, Cisco, NEC, 3Com, or any other major vendor's phones or platforms, then I may rethink my ideas. If 3Com and Digium are partnered up now, how come the NBX for V3000 doesn't support IAX2? They do have SIP. Second, there are work arounds for just about every downfall of SIP, like NAT traversal and the like. Third, ALL REAL TIME VOICE traffic is on a single port. There is a big issue there, I won't elaborate, but just think about it. SIP is here to stay until some other protocol comes about, but certainly not IAX2. It will be along the evolution of H323 to SIP to X., but not IAX,lol. Do you realize that most providers are dropping IAX2 support, even IAX.cc recommends SIP, gotta wonder why? Maybe it is all good now, but I won't bank my reputation on it. I use what I know works well, period. Even unnamed Digium Employees have poo pooed IAX2, albeit a year or two ago. It looks good on paper, didn't perform well historically, and now just like anything that I have lost trust in, it has to earn my trust back and that is not easy. I think a more useful thing to push for or put effort into is making Speex an industry standard codec. Now that would make alot of sense for everybody. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
