[asterisk-users] Asterisk Registrar / Trunk
Dears, 1-I have a GSM gateway (GOIP) with 8 ports, I used to let every port register to VoIPSwitch in order to know how many minutes does this GSM card, ASR ,ACD on each card. It's too simple on VoIPSwitch to add the registrar client to dial plan ,but in asterisk only I can find trunks How can I do that with asterisk . 2-Do any one know from where I can download a2billing prompts in Arabic for free. Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: mailto:kche...@xplorium.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com attachment: winmail.dat-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Speed Dials Management....
Hi Can you please send me a copy of the AGI script you wrote, in order to have look on it, it seems this is a solution for my problem Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William Stillwell Sent: Tuesday, August 16, 2011 5:02 PM To: sha...@a1telecoms.co.za; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Speed Dials Management What I have done is created a special extension # (ie, 63XXX) and then created a mysql database with XXX and the number to call, then when the 63xxx extension is dialed it looks up the number in the database via agi script and completes the call. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Shaun Wingrin Sent: Tuesday, August 16, 2011 4:35 AM To: asterisk-users Subject: [asterisk-users] Speed Dials Management Say, Is there any existing add-on / code etc. that manages speed dials. I find myself dialing number repeatedly and think that it would be great to have a system that can be controlled from the telephone instrument and work on the fly to build up a speed dial list. I would like that after I dial a number I can record a tag and have a speed dial no assigned. I should be able to dial using this speed dial and hear my tag played for me and also have the option of keying in a description. It would be great to be able to print this list of speed dials and the no's assigned to them. I use the TrixBox implementation... This is the closest I've found to what I'm looking for: http://www.ietf.org/rfc/rfc3398.txt http://www.ietf.org/rfc/rfc3398.txt Tx Shaun * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * image001.png-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AGI dialplan
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Re: [asterisk-users] Asterisk call limitation
Any update ? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W. Chehab Sent: Tuesday, June 21, 2011 12:40 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Asterisk call limitation The problem remains even when I add to /etc/init.d/asterisk ulimit -n 65536 [root@localhost ~]# ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 65536 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 1024 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 65536 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited [root@localhost ~]# -Original Message- From: Khaled W. Chehab [mailto:kche...@xplorium.com] Sent: Tuesday, June 21, 2011 12:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk call limitation Can you please specify more 1-how to set the ulimit on [root@localhost ~]# ulimit unlimited [root@localhost ~]# ulimit --help -bash: ulimit: --: invalid option ulimit: usage: ulimit [-SHacdfilmnpqstuvx] [limit] - How to set the ulimit command on in /etc/init.d/asterisk Since there is no parameter for ulimit in the file Thanks in advance Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Tuesday, June 21, 2011 12:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk call limitation Oh! Wait you set ulimit for running shellYou should set ulimit on asterisk. Also you can set ulimit command on asterisk startup file / etc/init.d/asterisk and restart asterisk also you can set in limit.conf file I had this issue before and I solved that way. -- Sent from my iPhone On Jun 20, 2011, at 4:47 PM, Khaled W. Chehab kche...@xplorium.com wrote: I tried the ulimit [root@localhost ~]# ulimit Unlimited Then sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) 100
[asterisk-users] Asterisk call limitation
Dears, i am using sipp to test asterisk(1.6.22) performance ,but when i limit the calls to 150 ,only 100 active calls on asterisk found ?why sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: mailto:kche...@xplorium.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com attachment: winmail.dat-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk call limitation
I tried the ulimit [root@localhost ~]# ulimit Unlimited Then sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss-noservice) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) 100 active channels 100 active calls 6407 calls processed [root@localhost ~]# I find in /var/log/asterisk/full [Jun 20 09:43:17] NOTICE[9705] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Jun 20 09:43:17] VERBOSE[3068] chan_unistim.c: Reloading unistim.conf... [Jun 20 16:43:33] WARNING[12353] file.c: Failed to write frame [Jun 20 16:43:34] WARNING[12389] file.c: Failed to write frame [Jun 20 16:43:35] WARNING[12394] file.c: Failed to write frame [Jun 20 16:43:43] WARNING[12484] file.c: Failed to write frame [Jun 20 16:43:44] WARNING[12488] file.c: Failed to write frame [Jun 20 16:43:52] WARNING[12573] file.c: Failed to write frame [Jun 20 16:43:57] WARNING[12625] file.c: Failed to write frame [Jun 20 16:44:07] WARNING[12723] file.c: Failed to write frame [Jun 20 16:44:14] WARNING[12789] file.c: Failed to write frame [Jun 20 16:44:22] WARNING[12872] file.c: Failed to write frame [Jun 20 16:44:26] WARNING[12908] file.c: Failed to write frame Khaled Chehab NGN Eng. Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Monday, June 20, 2011 11:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk call limitation It could be your OS limit try ulimit command. -- Sent from my iPhone On Jun 20, 2011, at 2:21 PM, Kevin P. Fleming kpflem...@digium.com wrote: On 06/20/2011 01:09 PM, Khaled W. Chehab wrote: Dears, i am using sipp to test asterisk(1.6.22) performance ,but when i limit the calls to 150 ,only 100 active calls on asterisk found ?why sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 You did not provide any log output, or anything that could be used to try to help you understand your problem. Without any details, any reply you get would be just a guess, nothing more. Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail:mailto:kche...@xplorium.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com Please refrain from including 20-line signature blocks in your messages
Re: [asterisk-users] Asterisk call limitation
Can you please specify more 1-how to set the ulimit on [root@localhost ~]# ulimit unlimited [root@localhost ~]# ulimit --help -bash: ulimit: --: invalid option ulimit: usage: ulimit [-SHacdfilmnpqstuvx] [limit] - How to set the ulimit command on in /etc/init.d/asterisk Since there is no parameter for ulimit in the file Thanks in advance Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Tuesday, June 21, 2011 12:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk call limitation Oh! Wait you set ulimit for running shellYou should set ulimit on asterisk. Also you can set ulimit command on asterisk startup file / etc/init.d/asterisk and restart asterisk also you can set in limit.conf file I had this issue before and I solved that way. -- Sent from my iPhone On Jun 20, 2011, at 4:47 PM, Khaled W. Chehab kche...@xplorium.com wrote: I tried the ulimit [root@localhost ~]# ulimit Unlimited Then sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) 100 active channels 100 active calls 6407 calls processed [root@localhost ~]# I find in /var/log/asterisk/full [Jun 20 09:43:17] NOTICE[9705] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Jun 20 09:43:17] VERBOSE[3068] chan_unistim.c: Reloading unistim.conf... [Jun 20 16:43:33] WARNING[12353] file.c: Failed to write frame [Jun 20 16:43:34] WARNING[12389] file.c: Failed to write frame [Jun 20 16:43:35] WARNING[12394] file.c: Failed to write frame [Jun 20 16:43:43] WARNING[12484] file.c: Failed to write frame [Jun 20 16:43:44] WARNING[12488] file.c: Failed to write frame [Jun 20 16:43:52] WARNING[12573] file.c: Failed to write frame [Jun 20 16:43:57] WARNING[12625] file.c: Failed to write frame [Jun 20 16:44:07] WARNING[12723] file.c: Failed to write frame [Jun 20 16:44:14] WARNING[12789] file.c: Failed to write frame [Jun 20 16:44:22] WARNING[12872] file.c: Failed to write frame [Jun 20 16:44:26] WARNING[12908] file.c: Failed to write frame Khaled Chehab NGN Eng. Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun
Re: [asterisk-users] Asterisk call limitation
The problem remains even when I add to /etc/init.d/asterisk ulimit -n 65536 [root@localhost ~]# ulimit -a core file size (blocks, -c) 0 data seg size (kbytes, -d) unlimited scheduling priority (-e) 0 file size (blocks, -f) unlimited pending signals (-i) 65536 max locked memory (kbytes, -l) 32 max memory size (kbytes, -m) unlimited open files (-n) 1024 pipe size(512 bytes, -p) 8 POSIX message queues (bytes, -q) 819200 real-time priority (-r) 0 stack size (kbytes, -s) 10240 cpu time (seconds, -t) unlimited max user processes (-u) 65536 virtual memory (kbytes, -v) unlimited file locks (-x) unlimited [root@localhost ~]# -Original Message- From: Khaled W. Chehab [mailto:kche...@xplorium.com] Sent: Tuesday, June 21, 2011 12:25 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Asterisk call limitation Can you please specify more 1-how to set the ulimit on [root@localhost ~]# ulimit unlimited [root@localhost ~]# ulimit --help -bash: ulimit: --: invalid option ulimit: usage: ulimit [-SHacdfilmnpqstuvx] [limit] - How to set the ulimit command on in /etc/init.d/asterisk Since there is no parameter for ulimit in the file Thanks in advance Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Tuesday, June 21, 2011 12:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk call limitation Oh! Wait you set ulimit for running shellYou should set ulimit on asterisk. Also you can set ulimit command on asterisk startup file / etc/init.d/asterisk and restart asterisk also you can set in limit.conf file I had this issue before and I solved that way. -- Sent from my iPhone On Jun 20, 2011, at 4:47 PM, Khaled W. Chehab kche...@xplorium.com wrote: I tried the ulimit [root@localhost ~]# ulimit Unlimited Then sipp -sn uac -d 1 -s 2005 127.0.0.1 -l 150 SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:4 Up Playback(ss- noservice) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) SIP/127.0.0.1:5061-0 s@from-trunk:5 Up SayAlpha(2005) 100 active channels 100 active calls 6407 calls processed [root@localhost ~]# I find in /var/log/asterisk/full [Jun 20 09:43:17] NOTICE[9705] pbx_ael.c: AEL load process: verified config file name '/etc/asterisk/extensions.ael'. [Jun 20 09:43:17] VERBOSE[3068] chan_unistim.c: Reloading unistim.conf... [Jun 20
[asterisk-users] Asterisk users Calculation
Dears I already read most of post on asterisk group and (http://www.voip-info.org/tiki-index.php?page=Asterisk+dimensioning) But I could not find a calculator 1-Is there a calculator I can download for that 2-What I the maximum simultaneous calls that can asterisk handle using CPU 3.0 MHZ and 4GB ram With rtp g729 and there is no codec transcoding , 3-And what is the number of simultaneous calls if I use direct RTP (Canreinvite=no /Directrt=yes) Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: mailto:kche...@xplorium.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com image001.png-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Best Scripting Language
perl libraries are so fast to manage/debug and easy to use,more over you can call too many function from system, and its good documented . Perl is the best J Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Satish Patel Sent: Friday, April 01, 2011 3:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Best Scripting Language No doubt perl is best. But python getting more popular these days. -- Sent from my iPhone On Apr 1, 2011, at 8:00 AM, mahesh katta maheshka...@flexydial.com wrote: Perl is the best script On Fri, Apr 1, 2011 at 5:27 PM, Gopalakrishnan A.N sai...@gmail.com wrote: Hi, Can anyone suggest which is the best scripting language for Asterisk or any telecom device? Thanks in advance. -- Thank you with regards, Gopalakrishnan A.N. VoIP call - sip:sai...@gtalk2voip.com mailto:sip%3asai...@gtalk2voip.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best Regards, Mahesh Katta BUZZWORKS Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI 303, Gagangiri Apts, Parleshwar Road, Ville Parle East, Mumbai - 400057. GSM +91.97029.70779 | Phone +91.22.2663.1811 | Fax +91.22.2663.1811 Web http://www.buzzworks.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * image001.png-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Skype
Can anyone make it more clear please
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: kche...@xplorium.com
MSN ID :khalidche...@hotmail.com
Web Site: http://www.xplorium.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Friday, February 25, 2011 11:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk/Skype
AFAIK, the issue here is not Skype or Gtalk. The Asterisk client isn't
really designed to easily transport messages during the call or otherwise.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell
Sent: Friday, February 25, 2011 3:14 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk/Skype
I am assuming that goes the same for Gtalk chat messages too?
Or has nobody played with that?
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Wilson
Sent: Friday, February 25, 2011 3:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk/Skype
On Feb 25, 2011, at 2:06 PM, Khaled W. Chehab wrote:
There is no debug appears,
Even I set core set verbose to 9
And skype set debug on
And in the extensions.conf I used
[Account]
exten = s,1,Set(message=${SKYPE_CHAT_RECEIVE(k_chehab,fakhourypbx,30)})
exten = s,n,NoOp(Received message: ${message})
The dialplan application is only for receiving chat messages during an
actual call. If you want to receive messages from outside of a call, you
will have to use the manager interface and look for SkyeChatMessage events.
image001.png--
_
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[asterisk-users] Asterisk/Skype
i installed skype for asterisk i can send and recieve calls normaly how can i receive messages from another skype user i Succeed to send only using for example: exten = 2233,1,SkypeChatSend(fSkypeBcp,User,message text) how to receive messages using this code SKYPE_CHAT_RECEIVE(account,from,timeout),and where and how I should add this code in extensions.conf my chan_Skype.conf [Account] secret=XX context=from-pstn exten= Account disallow=all allow=g729 allow=alaw allow=slin allow=ulaw auth_policy=accept buddy_presence=yes direction=both ;auth_policy=ignore buddy_autoadd=true ;buddy_presence=no mohinterpret=default ;mohsuggest=none Regards Khaled Chehab NGN Eng. Description: xplorium Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: mailto:kche...@xplorium.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com image001.png-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk/Skype
There is no debug appears,
Even I set core set verbose to 9
And skype set debug on
And in the extensions.conf I used
[Account]
exten = s,1,Set(message=${SKYPE_CHAT_RECEIVE(k_chehab,fakhourypbx,30)})
exten = s,n,NoOp(Received message: ${message})
any idea
regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: kche...@xplorium.com
MSN ID :khalidche...@hotmail.com
Web Site: http://www.xplorium.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of William
Stillwell
Sent: Friday, February 25, 2011 9:39 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Asterisk/Skype
Maybe something like this?
[skype_chat_receieve]
Exten = account,user,1,do something here?
What do you see in the CLI on the incoming txt message?
I just figured out how to handle a different google talk account today
[google-in]
Exten = us...@gmail.com,1,Dial(SIP/100)
Exten = us...@gmail.com,1,Dial(SIP/101)
Exten = us...@gmail.com,1,Dial(SIP/102)
It doesn't matter the context in gtalk or jingle ,..
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Friday, February 25, 2011 2:30 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Asterisk/Skype
i installed skype for asterisk
i can send and recieve calls normaly
how can i receive messages from another skype user
i Succeed to send only
using for example: exten = 2233,1,SkypeChatSend(fSkypeBcp,User,message
text)
how to receive messages using this code
SKYPE_CHAT_RECEIVE(account,from,timeout),and where and how I should
add this code in extensions.conf
my chan_Skype.conf
[Account]
secret=XX
context=from-pstn
exten= Account
disallow=all
allow=g729
allow=alaw
allow=slin
allow=ulaw
auth_policy=accept
buddy_presence=yes
direction=both
;auth_policy=ignore
buddy_autoadd=true
;buddy_presence=no
mohinterpret=default
;mohsuggest=none
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: kche...@xplorium.com
MSN ID :khalidche...@hotmail.com
Web Site: http://www.xplorium.com
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Re: [asterisk-users] Asterisk/Skype
Can you please send me a how to please or a simple lines?
Regards
Khaled Chehab
NGN Eng.
Description: xplorium
Operations Office - Lebanon
Office : +961 1 868686 ext 115
Mobile: +961 3 045212
E-mail: mailto:kche...@xplorium.com kche...@xplorium.com
MSN ID :khalidche...@hotmail.com
Web Site: http://www.xplorium.com
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Terry Wilson
Sent: Friday, February 25, 2011 10:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk/Skype
On Feb 25, 2011, at 2:06 PM, Khaled W. Chehab wrote:
There is no debug appears,
Even I set core set verbose to 9
And skype set debug on
And in the extensions.conf I used
[Account]
exten = s,1,Set(message=${SKYPE_CHAT_RECEIVE(k_chehab,fakhourypbx,30)})
exten = s,n,NoOp(Received message: ${message})
The dialplan application is only for receiving chat messages during an
actual call. If you want to receive messages from outside of a call, you
will have to use the manager interface and look for SkyeChatMessage events.
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Re: [asterisk-users] Asterisk Using as a SIP Client
Install asterisknow and begin from there. http://www.asterisk.org/asterisknow/ and don’t miss to read the documentation https://wiki.asterisk.org/wiki/display/AST/Home Regards Khaled Chehab NGN Eng. Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.xplorium.com -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nikhil Sent: Thursday, February 17, 2011 9:16 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Asterisk Using as a SIP Client Hi I wanted to use asterisk as SIP client in my centOS box.I should able to make calls and receive calls.and should able to talk and listen from the headset that I connected to my CentOS box. I need a direction to start on this. Thanks Nikhil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MOH RBT problem
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Re: [asterisk-users] Attack problem
Ircd is not installed and cant be located in all system ,any one know or have an idea how do they infect my system, Any bug in asterisknow? How to find the script that initiates this invites ? 135.307281 192.168.138.56 - 218.75.79.17 TCP 36578 ircd [ACK] Seq=36 Ack=111 Win=5840 Len=0 135.307434 192.168.138.56 - 218.75.79.17 TCP 36578 ircd [FIN, ACK] Seq=36 Ack=111 Win=5840 Len=0 135.309188 218.75.79.17 - 192.168.138.56 TCP ircd 36578 [FIN, ACK] Seq=111 Ack=1 Win=4096 Len=0 135.309211 192.168.138.56 - 218.75.79.17 TCP 36578 ircd [ACK] Seq=37 Ack=112 Win=5840 Len=0 135.334037 192.168.138.56 - 192.168.5.2 DNS Standard query A irc3.mysteryaddict.com 135.334496 192.168.5.2 - 192.168.138.56 DNS Standard query response A 87.229.45.226 135.334657 192.168.138.56 - 87.229.45.226 TCP 53718 ircd [SYN] Seq=0 Win=5840 Len=0 MSS=1460 TSV=1532274 TSER=0 WS=7 135.342359 218.75.79.17 - 192.168.138.56 TCP ircd 42802 [SYN, ACK] Seq=0 Ack=1 Win=1460 Len=0 MSS=1380 135.342399 192.168.138.56 - 218.75.79.17 TCP 42802 ircd [ACK] Seq=1 Ack=1 Win=5840 Len=0 135.342554 192.168.138.56 - 218.75.79.17 IRC Request Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A J Stiles Sent: Friday, December 17, 2010 6:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Attack problem On Friday 17 Dec 2010, Khaled W. Chehab wrote: HI, My system been attacked from someone I guess, kindly check the link below How can I stop the ircd attack # /etc/init.d/ircd stop # chmod -x /etc/init.d/ircd Should do the business :) -- AJS -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How to find , internal, external inbound or outbound
Hi, My system been attacked from someone I guess, kindly check the link below How can I stop the ircd attack http://pastebin.com/tbjh5qzP regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Attack problem
HI, My system been attacked from someone I guess, kindly check the link below How can I stop the ircd attack http://pastebin.com/tbjh5qzP regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax Degium channel License
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[asterisk-users] Inbound calls from TRUNK
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Re: [asterisk-users] Inbound calls from TRUNK
Thanks ,it solved by adding insecure=very regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W. Chehab Sent: Tuesday, September 28, 2010 2:16 PM To: Asterisk; Asterisk List Subject: [asterisk-users] Inbound calls from TRUNK Hi , I have configured sip trunk and put it in route for asterisk extensions How can I allow anonymous calls from trunk to extensions . All calls as a forbidden sip request Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial with MOH
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[asterisk-users] E1 R2 Congestion Status
I have a 'CONGESTION' Status with R2 protocol.
While testing this scenario sip GW--àAsterisk Digium E1 R2
ProtocolàCisco E1 R2 protocolàsip Gw
Find below my error and configuration ,where are the errors in my
configuration ?
=
Connected to Asterisk SVN-branch-1.6.2-r235775 currently running on
rev-212-98-156-56 (pid = 3614)
Verbosity is at least 3
== Using SIP RTP CoS mark 5
-- Executing [00223...@default:1] Dial(SIP/98.34.56.216-000e,
DAHDI/g1/00223344) in new stack
[Dec 22 06:02:49] WARNING[4756]: app_dial.c:1745 dial_exec_full: Unable to
create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion)
== Everyone is busy/congested at this time (1:0/1/0)
-- Auto fallthrough, channel 'SIP/98.34.56.216-000e' status is
'CONGESTION'
Cisco Gateway
show controller
Controller E1 slot(0)/port(0)
E1 Link is UP
No Alarm detected.
Applique type is Channelized E1.
Framing is CRC4, Line Code is HDB3.
Signalling type is R2-MFC.
0 Line Code Violations, 0 Framing Bit Errors
0 Far End Block Errors, 0 CRC Errors
signalling type = r2
clock source = slave
channel group 0 = 1-31
1 2 3
allocated timeslots = YYYNYYY
outgoing barred channel group =
channel order = ascending
b-channel negotiation = exclusive
overlap receiving by forced = disabled
overlap sending by forced = disabled
protocol side = network
R2 get calling number = none
ISDN virtual connect = disabled
ISDN Layer 2 is DOWN
ISDN Values
ISDN Layer 2 values
k= 7
N200 = 3
N201 = 260
T200 = 1 seconds
T203 = 10 seconds
ISDN Layer 3 values
T301 = 180 seconds
T303 = 4 seconds
T304 = 20 seconds
T305 = 30 seconds
T306 = 30 seconds
T308 = 4 seconds
T310 = 10 seconds
T313 = 10 seconds
T316 = 120 seconds
T322 = 4 seconds
T309 = 90 seconds
N303 = 1
---
/etc/asterisk/chan_dahdi.conf
[trunkgroups]
signalling=mfcr2
mfcr2_variant=mx
trunkgroup = 1,16
spanmap = 1,1,1
[channels]
signalling=mfcr2
mfcr2_variant=mx
context=default
signalling=mfcr2
mfcr2_variant=mx
signalling=mfcr2
mfcr2_variant=mx
usecallerid=yes
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
group=1
callgroup=1
pickupgroup=1
callerid = asreceived
useincomingcalleridondahditransfer = yes
tonezone = 0 ; 0 is US
channel = 1-15,17-31
signalling=mfcr2
mfcr2_variant=itu
mfcr2_max_ani=7
mfcr2_max_dnis=8
mfcr2_get_ani_first=no
mfcr2_category=national_subscriber
mfcr2_logdir=span1
mfcr2_logging=all
;EOF
cat /etc/dahdi/system.conf
# Autogenerated by /usr/sbin/dahdi_genconf on Tue Dec 22 01:59:02 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
#
# Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 RED
span=1,1,0,cas,hdb3,crc4
# termtype: te
#bchan=1-15,17-31
#dchan=16
cas=1-15:1101
dchan=16
cas=17-31:1101
echocanceller=mg2,1-15,17-31
# Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 RED
span=2,2,0,ccs,hdb3,crc4
# termtype: te
bchan=32-46,48-62
dchan=47
echocanceller=mg2,32-46,48-62
# Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 RED
span=3,3,0,ccs,hdb3,crc4
# termtype: te
bchan=63-77,79-93
dchan=78
echocanceller=mg2,63-77,79-93
# Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4 RED
span=4,4,0,ccs,hdb3,crc4
# termtype: te
bchan=94-108,110-124
dchan=109
echocanceller=mg2,94-108,110-124
# Global data
loadzone= us
defaultzone = us
[default]
exten = _X.,1,Dial(DAHDI/g1/${EXTEN})
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without express written confirmation
by an officer of Xplorium. Any views expressed by an individual in this
electronic message do not necessarily reflect views of Xplorium or its
subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
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If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other
[asterisk-users] CDR Import
Hi, how to write the cdr directly to the databse (Mysq)instead of importing Master.csv to table using a php script. Noting that I load asterisk_addons_mysql rev-xx-xx-xx-xx*CLI cdr status rev-xx-xx-xx-xx*CLI Call Detail Record (CDR) settings -- Logging:Enabled Mode: Simple Log unanswered calls: Yes * Registered Backends --- csv cdr_sqlite3_custom cdr-custom regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] E1 Extensions.conf
Hi, I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02)) 4 ports I want to make a loop test between digium card E1 to test the configuration of dahdi What I want to do scenario is I connect port 1 and port4 in the digium card with E1 cable SIPcall--E1 Digium port 1---(Loop)E1 port 2sip extension local. kindly can any can help me to draw this dialpan in the extensions.conf Description Alarms IRQbpviol CRC4 Fra Codi Options LBO T4XXP (PCI) Card 0 Span 1OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 2RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 3RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 4OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) Khaled Chehab NGN Eng. Untitled Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: mailto:bs...@mg-tel.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.Xplorium.com http://www.Xplorium.com * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * image001.jpg___ -- 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] E1 Extensions.conf
Find my dahdi config files below dahdi-channels.conf ; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 ClockSource group=0,11 context=default switchtype = euroisdn signalling = pri_cpe channel = 1-15,17-31 context = default group = 63 ; Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 RED group=0,12 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 32-46,48-62 context = default group = 63 ; Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 RED group=0,13 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 63-77,79-93 context = default group = 63 ; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4 group=0,14 context=from-pstn switchtype = euroisdn signalling = pri_cpe channel = 94-108,110-124 context = default group = 63 Chan_dahdi.conf [trunkgroups] [channels] language=en context=default signalling = pri_cpe callwaiting=yes hidecallerid=no usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes echocancel=no echocancelwhenbridged=no relaxdtmf=yes usedistinctiveringdetection=yes usecallingpres=yes busydetect=yes callprogress=yes rxgain=2.0 txgain=2.0 #include dahdi-channels.conf /etc/dahdi/system.conf # Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER) HDB3/CCS/CRC4 ClockSource span=1,1,0,ccs,hdb3,crc4 # termtype: te bchan=1-15,17-31 dchan=16 echocanceller=mg2,1-15,17-31 # Span 2: TE4/0/2 T4XXP (PCI) Card 0 Span 2 HDB3/CCS/CRC4 RED span=2,2,0,ccs,hdb3,crc4 # termtype: te bchan=32-46,48-62 dchan=47 echocanceller=mg2,32-46,48-62 # Span 3: TE4/0/3 T4XXP (PCI) Card 0 Span 3 HDB3/CCS/CRC4 RED span=3,3,0,ccs,hdb3,crc4 # termtype: te bchan=63-77,79-93 dchan=78 echocanceller=mg2,63-77,79-93 # Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4 HDB3/CCS/CRC4 span=4,4,0,ccs,hdb3,crc4 # termtype: te bchan=94-108,110-124 dchan=109 echocanceller=mg2,94-108,110-124 # Global data loadzone= us defaultzone = us Hi, I have a digium card (igium, Inc. Wildcard TE405P quad-span T1/E1/J1 card 5.0V (rev 02)) 4 ports I want to make a loop test between digium card E1 to test the configuration of dahdi What I want to do scenario is I connect port 1 and port4 in the digium card with E1 cable SIPcall--E1 Digium port 1---(Loop)E1 port 2sip extension local. kindly can any can help me to draw this dialpan in the extensions.conf Description Alarms IRQbpviol CRC4 Fra Codi Options LBO T4XXP (PCI) Card 0 Span 1OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 2RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 3RED 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) T4XXP (PCI) Card 0 Span 4OK 0 0 0 CCS HDB3 CRC4 0 db (CSU)/0-133 feet (DSX-1) Khaled Chehab NGN Eng. Untitled Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com mailto:bs...@mg-tel.com MSN ID :khalidche...@hotmail.com Web Site: http://www.Xplorium.com _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium.
Re: [asterisk-users] E1 Extensions.conf
Hi,
I have a digium card (digium, Inc. Wildcard TE405P quad-span T1/E1/J1 card
5.0V (rev 02)) 4 ports
I want to make a loop test between spans on digium card in order to test
the spans.
I connect port 1 and port4 with cross E1 cable
I am trying to do this scenario
SIPcall-- Digium span 1---(Loop)Span 4sip
mailto:extens...@xx.xx.xx.xx extens...@xx.xx.xx.xx.
Kindly can you help me on how to forward the call from Span1-àSpan4 and then
from span4-à...@xx.xx.xx
My dahdi_channels.conf
; Span 1: TE4/0/1 T4XXP (PCI) Card 0 Span 1 (MASTER)
group=1
context=default
switchtype = euroisdn
signalling = pri_net
channel = 1-15,17-31
context = default
;group = 63
; Span 4: TE4/0/4 T4XXP (PCI) Card 0 Span 4
group=4
;context=default
switchtype = euroisdn
signalling = pri_cpe
channel = 94-108,110-124
context = incomingck
;group = 63
-extensions.conf-
[default]
exten = _X.,1,Dial(DAHDI/G1/${EXTEN})
[incomingck]
exten = _X.,1,Dial(SIP/96123...@212.98.141.217,60)
Regards
*
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[asterisk-users] Asterisk SS7 Sigtran Protocol
Dears, Do Asterisk support SS7 SIGTRAN(SS7 over IP) protocol ? And how to integrate Regards Khaled Chehab NGN Eng. Untitled Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: mailto:bs...@mg-tel.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.Xplorium.com http://www.Xplorium.com * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * image001.jpg___ -- 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 Fax Module
When we can expect to have a res_fax and res_fax_degium module for asterisk V 1.6.2 Regards Khaled Chehab NGN Eng. Untitled Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com mailto:bs...@mg-tel.com MSN ID :khalidche...@hotmail.com Web Site: http://www.Xplorium.com * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * image001.jpg___ -- 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] G729
I have problemin g729 codec compatibility,I get the g729 module from http://asterisk.hosting.lv/ and I have Asterisk 1.4.22-3 RPM What g729 module should I download ? I already downloaded codec_g723-ast14-icc-glibc-pentium4.so [trixbox1.localdomain asterisk]# cat /proc/cpuinfo processor : 0 vendor_id : GenuineIntel cpu family : 15 model : 4 model name : Intel(R) Xeon(TM) CPU 3.40GHz stepping: 1 cpu MHz : 3399.733 cache size : 1024 KB fdiv_bug: no hlt_bug : no f00f_bug: no coma_bug: no fpu : yes fpu_exception : yes cpuid level : 5 wp : yes flags : fpu vme de pse tsc msr pae mce cx8 apic mtrr pge mca cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss constant_tsc up pni bogomips: 6813.20 please advice regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] (no subject)
Hi
I use dial with music on hold command
exten = _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem
if the called party line is closed or number is incorrect or have a voice
mail (Early media 183) user will not hear the message from operator
notifying that line is out of service , temporary unavailable .,
what to do to solve this problem
In other words how to stop MOH since asterisk detect 183 and even if i can
do that when the 183 comes from my soft switch which will allow user to hear
the Ring Back Tone
i found in the app_dial.c
case AST_CONTROL_RINGING:
Thanks in advance
*
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[asterisk-users] 183 early media
HI all ,
I am using ,Dial(SIP/Gateway/${EXTEN},m)
how can i modify asterisk, if it detects two early media to stop OR MUTE
the first RTP early media AND let the user hear the second early media
any one developed something like that or know from where I can do this from
chan_sip.c?
regards
*
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[asterisk-users] Fax t38 capability
Dears I installed digium fax and followed the instruction at
http://downloads.digium.com/pub/telephony/fax/README,And as you can see
above that t38 is loaded
I am using a call file to send fax1.tif file as fax to the gateway named
add
The problem that Addpac send always Receive 488 Not acceptable here,and
lkindly find my debug attached
Please advice.
Thanks I Advance
shark*CLI fax show capabilities
shark*CLI
Registered Fax Technology Modules:
Type: T.38
Description : Digium Fax T.38 Driver
Capabilities: SEND, RECEIVE, UDP
Type: G.711
Description : Digium Fax G.711 Driver
Capabilities: SEND, RECEIVE
2 registered modules
My Call File***
Channel: SIP/6...@add
MaxRetries: 2
WaitTime: 20
Extension: s
Priority: 1
Context: fax-tx
*My sipconf**
Cisco Gateway
[add]
host=212.56.151.216
username=***userid***
secret=***password***
type=peer
t38pt_udptl = yes
*My extension.conf**
[fax-tx]
exten = s,1,NoOp( SENDING FAX )
exten = s,n,Wait(6)
exten = s,n,Set(GLOBAL(FAXCOUNT)=$[ ${GLOBAL(FAXCOUNT)} + 1 ])
exten = s,n,Set(FAXCOUNT=${GLOBAL(FAXCOUNT)})
exten = s,n,Set(FAXFILE=fax1.tif)
; Set FAXOPTs
exten = s,n,NoOp( SETTING FAXOPT )
exten = s,n,Set(FAXOPT(filename)=${FAXFILE})
exten = s,n,Set(FAXOPT(ecm)=yes)
exten = s,n,Set(FAXOPT(headerinfo)=Fax from ${GLOBAL(LASTFAXCALLERNAME)} at
${GLOBAL(LASTFAXCALLERNUM)} was received.)
exten = s,n,Set(FAXOPT(localstationid)=1234567890)
exten = s,n,Set(FAXOPT(maxrate)=14400)
exten = s,n,Set(FAXOPT(minrate)=2400)
; Send the fax
exten = s,n,NoOp( SENDING FAX : ${FAXFILE} )
exten = s,n,SendFAX(/tmp/${FAXFILE},d)
; Hangup! Print FAXOPTs
exten = h,1,NoOp(FAXOPT(ecm) : ${FAXOPT(ecm)})
exten = h,n,NoOp(FAXOPT(filename) : ${FAXOPT(filename)})
exten = h,n,NoOp(FAXOPT(headerinfo) : ${FAXOPT(headerinfo)})
exten = h,n,NoOp(FAXOPT(localstationid) : ${FAXOPT(localstationid)})
exten = h,n,NoOp(FAXOPT(maxrate) : ${FAXOPT(maxrate)})
exten = h,n,NoOp(FAXOPT(minrate) : ${FAXOPT(minrate)})
exten = h,n,NoOp(FAXOPT(pages) : ${FAXOPT(pages)})
exten = h,n,NoOp(FAXOPT(rate) : ${FAXOPT(rate)})
exten = h,n,NoOp(FAXOPT(remotestationid) : ${FAXOPT(remotestationid)})
exten = h,n,NoOp(FAXOPT(resolution) : ${FAXOPT(resolution)})
exten = h,n,NoOp(FAXOPT(status) : ${FAXOPT(status)})
exten = h,n,NoOp(FAXOPT(statusstr) : ${FAXOPT(statusstr)})
exten = h,n,NoOp(FAXOPT(error) : ${FAXOPT(error)})
*
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***Debug
shark*CLI
Fax Debug Enabled
-- Attempting call on SIP/6...@add for 1...@fax-tx:1 (Retry 1)
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Channel SIP/add-084acf98 was answered.
== Starting SIP/add-084acf98 at fax-tx,123,1 failed so falling back to exten
's'
-- Executing [...@fax-tx:1] NoOp(SIP/add-084acf98, SENDING FAX
) in new stack
-- Executing [...@fax-tx:2] Wait(SIP/add-084acf98, 6) in new stack
-- Executing [...@fax-tx:3] Set(SIP/add-084acf98, GLOBAL(FAXCOUNT)=) in
new stack
== Setting global variable 'FAXCOUNT' to ''
-- Executing [...@fax-tx:4] Set(SIP/add-084acf98, FAXCOUNT=) in new
stack
-- Executing [...@fax-tx:5] Set(SIP/add-084acf98, FAXFILE=fax1.tif) in
new stack
-- Executing [...@fax-tx:6] NoOp(SIP/add-084acf98, SETTING FAXOPT
) in new stack
-- Executing [...@fax-tx:7] Set(SIP/add-084acf98,
FAXOPT(filename)=fax1.tif) in new stack
-- Executing [...@fax-tx:8] Set(SIP/add-084acf98, FAXOPT(ecm)=yes) in
new stack
-- Executing [...@fax-tx:9] Set(SIP/add-084acf98, FAXOPT(headerinfo)=Fax
from at was received.) in new stack
-- Executing [...@fax-tx:10] Set(SIP/add-084acf98,
FAXOPT(localstationid)=1234567890) in new stack
-- Executing [...@fax-tx:11]
[asterisk-users] Dial with MOH
Hi
I use dial with music on hold command
exten = _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem
if the called party line is closed or number is incorrect or have a voice
mail (Early media 183) user will not hear the message from operator
notifying that line is out of service , temporary unavailable .,
what to do to solve this problem
Thank you in advance.
*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without express written confirmation
by an officer of Xplorium. Any views expressed by an individual in this
electronic message do not necessarily reflect views of Xplorium or its
subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message and any of
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[asterisk-users] stop the MOH since asterisk knows that channel is ringing
Hi
I use dial with music on hold command
exten = _X.,n,Dial(SIP/Trunk/${EXTEN}|300|m),I am facing a big problem
if the called party line is closed or number is incorrect or have a voice
mail (Early media 183) user will not hear the message from operator
notifying that line is out of service , temporary unavailable .,
what to do to solve this problem
In other words how to stop MOH since asterisk detect 183 and even if i can
do that when the 183 comes from my soft switch which will allow user to hear
the Ring Back Tone
i found in the app_dial.c
case AST_CONTROL_RINGING:
Thanks in advance
*
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of Xplorium with another party by e-mail without express written confirmation
by an officer of Xplorium. Any views expressed by an individual in this
electronic message do not necessarily reflect views of Xplorium or its
subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message and any of
its attachments, or that they are free from computer viruses or other defects.
*
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[asterisk-users] Channel/Mute
Hi.
Does asterisk support muting per a specific channel?
(like the soft hangup command, were you specify a channel and then
asterisks hangs it up).
1-If it does not, how will one go about to do something like this?
2-how to let the user hear 183 the early media like voice mail prompt since
I am using MOH when dialing
Dial(SIP/${EXTEN},30,m)
Thank you in advance.
*
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subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
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Xplorium does not guarantee the integrity of this electronic message and any of
its attachments, or that they are free from computer viruses or other defects.
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[asterisk-users] Asterisk Double Invite
Dears My scenario is incoming call to asterisk which asterisk in its term will dial it through its trunk . I recognized that Asterisk is sending two invites to My Trunk GW IP as you can see in the debugging below The first is the default and the second when asterisk receives a 200 OK Why Asterisk(B2BUA) is acting like that, and from where I can get the asterisk sip dial call flow Why Asterisk is sending double invite GW CLIENT IP=192.168.5.100 Asterisk IP=192.168.5.150 Termination GW=192.168.5.200 Please find sip debug at http://pastebin.com/m6e7f454 http://pastebin.com/m6e7f454 and the tethereal below Capturing on eth0 4.865698 192.168.5.100- 192.168.5.150 SIP/SDP Request: INVITE sip:3316234335...@192.168.5.150, with session description 4.871457 192.168.5.150 - 192.168.5.100SIP Status: 100 Trying 4.876797 192.168.5.150 - 192.168.5.100SIP/SDP Status: 183 Session Progress, with session description 6.947270 192.168.5.150 - 192.168.5.200 SIP/SDP Request: INVITE sip:3316234335...@192.168.5.200, with session description 6.949157 192.168.5.200 - 192.168.5.150 SIP Status: 100 Trying 12.759311 192.168.5.200 - 192.168.5.150 SIP/SDP Status: 183 Session Progress, with session description 16.236320 192.168.5.200 - 192.168.5.150 SIP Status: 180 Ringing 20.250002 192.168.5.200 - 192.168.5.150 SIP/SDP Status: 200 OK, with session description 20.250395 192.168.5.150 - 192.168.5.200 SIP Request: ACK sip:3316234335...@192.168.5.200:5060 20.251267 192.168.5.150 - 192.168.5.100SIP/SDP Status: 200 OK, with session description 20.251752 192.168.5.150 - 192.168.5.200 SIP/SDP Request: INVITE sip:3316234335...@192.168.5.200:5060, with session description 20.252986 192.168.5.200 - 192.168.5.150 SIP Status: 100 Trying 20.274788 192.168.5.200 - 192.168.5.150 SIP/SDP Status: 200 OK, with session description 20.275143 192.168.5.150 - 192.168.5.200 SIP Request: ACK sip:3316234335...@192.168.5.200:5060 20.569819 192.168.5.100- 192.168.5.150 SIP Request: ACK sip:3316234335...@192.168.5.150 20.570303 192.168.5.150 - 192.168.5.100SIP/SDP Request: INVITE sip:551130338...@192.168.5.100, with session description 20.900485 192.168.5.100- 192.168.5.150 SIP/SDP Status: 200 OK, with session description 20.902604 192.168.5.150 - 192.168.5.100SIP Request: ACK sip:551130338...@192.168.5.100 32.468119 192.168.5.200 - 192.168.5.150 SIP Request: BYE sip:551130338...@192.168.5.150 32.468411 192.168.5.150 - 192.168.5.200 SIP Status: 200 OK 32.468750 192.168.5.150 - 192.168.5.100SIP/SDP Request: INVITE sip:551130338...@192.168.5.100, with session description 32.822154 192.168.5.100- 192.168.5.150 SIP/SDP Status: 200 OK, with session description 32.822478 192.168.5.150 - 192.168.5.100SIP Request: ACK sip:551130338...@192.168.5.100 32.822928 192.168.5.150 - 192.168.5.100SIP Request: BYE sip:551130338...@192.168.5.100 33.140288 192.168.5.100- 192.168.5.150 SIP Status: 200 OK * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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 Double Invite
Dears My scenario is incoming call to asterisk which asterisk in its term will dial it through its trunk . I recognized that Asterisk is sending two invites to My Trunk GW IP as you can see in the debugging below The first is the default and the second when asterisk receives a 200 OK Why Asterisk(B2BUA) is acting like that, and from where I can get the asterisk sip dial call flow Why Asterisk is sending double invite GW CLIENT IP=192.168.5.100 Asterisk IP=192.168.5.150 Termination GW=192.168.5.200 Capturing on eth0 4.865698 192.168.5.100- 192.168.5.150 SIP/SDP Request: INVITE sip:3316234335...@192.168.5.150, with session description 4.871457 192.168.5.150 - 192.168.5.100SIP Status: 100 Trying 4.876797 192.168.5.150 - 192.168.5.100SIP/SDP Status: 183 Session Progress, with session description 6.947270 192.168.5.150 - 192.168.5.200 SIP/SDP Request: INVITE sip:3316234335...@192.168.5.200, with session description 6.949157 192.168.5.200 - 192.168.5.150 SIP Status: 100 Trying 12.759311 192.168.5.200 - 192.168.5.150 SIP/SDP Status: 183 Session Progress, with session description 16.236320 192.168.5.200 - 192.168.5.150 SIP Status: 180 Ringing 20.250002 192.168.5.200 - 192.168.5.150 SIP/SDP Status: 200 OK, with session description 20.250395 192.168.5.150 - 192.168.5.200 SIP Request: ACK sip:3316234335...@192.168.5.200:5060 20.251267 192.168.5.150 - 192.168.5.100SIP/SDP Status: 200 OK, with session description 20.251752 192.168.5.150 - 192.168.5.200 SIP/SDP Request: INVITE sip:3316234335...@192.168.5.200:5060, with session description 20.252986 192.168.5.200 - 192.168.5.150 SIP Status: 100 Trying 20.274788 192.168.5.200 - 192.168.5.150 SIP/SDP Status: 200 OK, with session description 20.275143 192.168.5.150 - 192.168.5.200 SIP Request: ACK sip:3316234335...@192.168.5.200:5060 20.569819 192.168.5.100- 192.168.5.150 SIP Request: ACK sip:3316234335...@192.168.5.150 20.570303 192.168.5.150 - 192.168.5.100SIP/SDP Request: INVITE sip:551130338...@192.168.5.100, with session description 20.900485 192.168.5.100- 192.168.5.150 SIP/SDP Status: 200 OK, with session description 20.902604 192.168.5.150 - 192.168.5.100SIP Request: ACK sip:551130338...@192.168.5.100 32.468119 192.168.5.200 - 192.168.5.150 SIP Request: BYE sip:551130338...@192.168.5.150 32.468411 192.168.5.150 - 192.168.5.200 SIP Status: 200 OK 32.468750 192.168.5.150 - 192.168.5.100SIP/SDP Request: INVITE sip:551130338...@192.168.5.100, with session description 32.822154 192.168.5.100- 192.168.5.150 SIP/SDP Status: 200 OK, with session description 32.822478 192.168.5.150 - 192.168.5.100SIP Request: ACK sip:551130338...@192.168.5.100 32.822928 192.168.5.150 - 192.168.5.100SIP Request: BYE sip:551130338...@192.168.5.100 33.140288 192.168.5.100- 192.168.5.150 SIP Status: 200 OK * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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 420 Bad Response
Dears, When my GW send a call to asterisk v 1.4.24 , Asterisk send Status: 420 bad extension (unsupported) Why? Any modifications should be done one sip.conf regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] Trunks
Dears How to disallow asterisk to send the keep alive 200 ok message to the peers and trunks. Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] MOH
Dears -How can I stop MOH when status of the dial is ringing and let the user hear the Ring Back Tone from the termination Gateway. Even I can see in the CLI debugging the status is ringing -my idea is to add music on hold stop when asterisk detect -- SIP/OPNS-096456c0 is ringing line In which script this line located? -- Executing [97130245...@default:1] SetMusicOnHold(SIP/xx.xx.xx.xx-096ca8c0, English) in new stack -- Executing [9713024...@default:2] Dial(SIP/xx.xx.xx.xx-096ca8c0, SIP/OPNS/9713024561|300|m) in new stack -- Called OPNS/9713024561 -- Started music on hold, class 'English', on SIP/xx.xx.xx.xx-096ca8c0 -- SIP/OPNS-096456c0 is ringing -More over when I used directrtpsetup=yes I heard the MOH and the ring back tone together . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] MOH
Dear Ben, I tried a lot ,Kindly can you give me an example on how to do that using a macro. Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Tuesday, April 14, 2009 5:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MOH Khaled W. Chehab wrote: Dears -How can I stop MOH when status of the dial is ringing and let the user hear the Ring Back Tone from the termination Gateway. Remove the 'm' out of your dial command: m([class]) - Provide hold music to the calling party until a requested channel answers. A specific MusicOnHold class can be specified. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] MOH
Thanks for answering Doug
I am using exten = _X.,n,Dial(SIP/OPNS/${EXTEN}|300|m) with no macros
kindly can you wrote down a macro to stop the MOH RTP in order to let the
GW inband early media rtp heard by the caller
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Tuesday, April 14, 2009 8:57 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MOH
Khaled W. Chehab wrote:
Dear Ben,
I tried a lot ,Kindly can you give me an example on how to do that using a
macro.
Remove the 'm' out of your dial command:
Doug
I'm not Ben, but I'll answer.
Shows us what your macro looks like and we'll chime in with some pointers.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
Dears
-How can I stop MOH when status of the dial is ringing and let the user hear
the Ring Back Tone from the termination Gateway.
Even I can see in the CLI debugging the status is ringing -my idea is to add
music on hold stop when asterisk detect -- SIP/OPNS-096456c0 is ringing
line
In which script this line located?
-- Executing [97130245...@default:1]
SetMusicOnHold(SIP/xx.xx.xx.xx-096ca8c0, English) in new stack
-- Executing [9713024...@default:2] Dial(SIP/xx.xx.xx.xx-096ca8c0,
SIP/OPNS/9713024561|300|m) in new stack
-- Called OPNS/9713024561
-- Started music on hold, class 'English', on SIP/xx.xx.xx.xx-096ca8c0
-- SIP/OPNS-096456c0 is ringing
-More over when I used directrtpsetup=yes I heard the MOH and the ring back
tone together .
Regards
*
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electronic message do not necessarily reflect views of Xplorium or its
subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
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Re: [asterisk-users] MOH
Man :)
I want the MOH play until Asterisk receives 180 ringing or 183 from the
termination GW.
Here I want to stop the MOH and let the user hear the early media RBT
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Tuesday, April 14, 2009 9:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MOH
Khaled W. Chehab wrote:
Thanks for answering Doug
I am using exten = _X.,n,Dial(SIP/OPNS/${EXTEN}|300|m) with no macros
Change this to:
_X.,n,Dial(SIP/OPNS/${EXTEN}|300)
The m was causing the music on hold.
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety.
___
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*
No employee or agent is authorized to conclude any binding agreement on behalf
of Xplorium with another party by e-mail without express written confirmation
by an officer of Xplorium. Any views expressed by an individual in this
electronic message do not necessarily reflect views of Xplorium or its
subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message and any of
its attachments, or that they are free from computer viruses or other defects.
*
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Re: [asterisk-users] MOH
Any idead on how to begin with AGI -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Tuesday, April 14, 2009 10:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MOH Khaled W. Chehab wrote: Man :) I want the MOH play until Asterisk receives 180 ringing or 183 from the termination GW. I don't think you'll be able to mix and match via the dial application. You may have to try using AGI for this. That, I can't help you with. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ -- 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 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] Please Advice SIP 183 progessl
Kindly can you send me the code ,or how to Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin Sent: Monday, April 06, 2009 7:13 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Please Advice SIP 183 progessl Hi, The easiest is to turn off MOH on the Dial. Otherwise the patch is easy but not trivial. Once the B-leg receives the ringing message and passes it in Dial app then the code has to turn off the MOH and tell the A-leg to send the ringing message. At the same time the code that skips passing the ringing to A-leg has to be disabled. Martin On Mon, Apr 6, 2009 at 2:38 AM, Khaled W. Chehab kche...@xplorium.com wrote: Dear Martin Can you inform me how to make the patch or from where I can get it otherwise if there is an application can generate it? Or if its relate to chan_sip.c ,please can you tell me which function to edit or lines to be added Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin Sent: Sunday, April 05, 2009 5:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Please Advice SIP 183 progessl Hi Khaled, app Dial clearly is coded to ignore the 180 Ringing being passed if you have 'm' option to Dial and you do. Try to take the 'm' out and see if 180 Ringing is passed to the A-leg. So if you want MOH and then when 180 Ringing comes to turn it off = you need a patch. Martin 2009/4/4 Khaled W. Chehab kche...@xplorium.com: 10x Martin , But B-Leg is sending 180 ringing 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 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] Please Advice SIP 183 progessl
Dear Martin Can you inform me how to make the patch or from where I can get it otherwise if there is an application can generate it? Or if its relate to chan_sip.c ,please can you tell me which function to edit or lines to be added Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin Sent: Sunday, April 05, 2009 5:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Please Advice SIP 183 progessl Hi Khaled, app Dial clearly is coded to ignore the 180 Ringing being passed if you have 'm' option to Dial and you do. Try to take the 'm' out and see if 180 Ringing is passed to the A-leg. So if you want MOH and then when 180 Ringing comes to turn it off = you need a patch. Martin 2009/4/4 Khaled W. Chehab kche...@xplorium.com: 10x Martin , But B-Leg is sending 180 ringing 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 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] Relay ringing sip message 180
Dears
Asterisk is a median server between the caller and the terminations gateway
The caller send the call to asterisk à asterisk will play music on hold
untill the termination gateway send 200 OK and the RTP establish
My problem that, Asterisk is not forwarding the 180 ringing from the
termination gateway to the user
How can I forward the sip message 180 to the caller or let the music on hold
stop playing and the caller hears the Ring Back Tone which when 180
ringing from the termination gateway.
What I am using now to stop the musinc on hold when the RTP established is
exten = _X.,n,Dial(SIP/Temination_Gateway/${EXTEN}|300|m)
NB: I tried to edit chan_sip.c but I did not find the solution
Please Advice
Regards
*
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Re: [asterisk-users] Please Advice SIP 183 progessl
10x Martin ,
But B-Leg is sending 180 ringing
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Martin
Sent: Saturday, April 04, 2009 9:09 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Hi Khaled,
I believe the 180 Ringing will be sent only if your B-leg sends it to
Asterisk.
Asterisk doesn't know WHEN the call will physically ring the destination
number
so unless you GW tells it you won't ever see that message unless you patch
it.
DISCLAIMER: I may be wrong and was wrong before.
Martin
On Thu, Apr 2, 2009 at 11:07 AM, Khaled W. Chehab kche...@xplorium.com
wrote:
Dears
Kindly find my dial script below,I am trying to send the caller 180
ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
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Re: [asterisk-users] Please Advice SIP 183 progessl
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
or if there any way to stop the music on hold and let the caller hear the
Ring Back Tone
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
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in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
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attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
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defects.
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Re: [asterisk-users] Xorcom and Doorbell
Dears How can I send or force sending 180 Ringing instead of 183 back to the caller ?or send both sequential if its impossible I used progressinband=never but it did work . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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 183 progessl
Can you please tell me how to Custom SIP header Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Thursday, April 02, 2009 6:16 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Xorcom and Doorbell Custom SIP header? -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W. Chehab Sent: Thursday, April 02, 2009 10:02 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Xorcom and Doorbell Dears How can I send or force sending 180 Ringing instead of 183 back to the caller ?or send both sequential if its impossible I used progressinband=never but it did work . Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] Please Advice SIP 183 progessl
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects.
*
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in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
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If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
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Xplorium does not guarantee the integrity
Re: [asterisk-users] Please Advice SIP 183 progessl
Guys I tried the R,r options at the DIAL(SIP/,,rm) and also its sending
183
Any Advice
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects.
*
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No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment
Re: [asterisk-users] Please Advice SIP 183 progessl
I tried it but it didn't work even ,If I use Answer() function , Billing
will starts
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 8:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
This is a hack-fix but if you Answer the call before dialing, that might
remove the progress message
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 12:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Guys I tried the R,r options at the DIAL(SIP/,,rm) and also its sending
183
Any Advice
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
-- add --
Exten = _X.,n,Answer()
-- end add --
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments, kindly delete it immediately from your system and notify the
sender by electronic mail. You must not copy this message or attachment or
disclose its content to any other person.
Xplorium does not guarantee the integrity of this electronic message and any
of its attachments, or that they are free from computer viruses or other
defects.
*
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Re: [asterisk-users] Please Advice SIP 183 progessl
Do you know how to play a musiconhold or ... but when its ringing the user
will hear the Ring Back Tone
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 9:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Try replacing answer() with playback(tt-monkeys)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 1:33 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
I tried it but it didn't work even ,If I use Answer() function , Billing
will starts
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 8:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
This is a hack-fix but if you Answer the call before dialing, that might
remove the progress message
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 12:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Guys I tried the R,r options at the DIAL(SIP/,,rm) and also its sending
183
Any Advice
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
-- add --
Exten = _X.,n,Answer()
-- end add --
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential information protected from disclosure
belonging to Xplorium.
If you are not the intended addressee of this electronic message and its
attachments
Re: [asterisk-users] Please Advice SIP 183 progessl
Kindly its too important to me
If any one can help me on a command can force asterisk to send 180 and 183
sip message in the same time
Regards
Do you know how to play a musiconhold or ... but when its ringing the user
will hear the Ring Back Tone
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 9:44 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Try replacing answer() with playback(tt-monkeys)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 1:33 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
I tried it but it didn't work even ,If I use Answer() function , Billing
will starts
Thanks
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 8:59 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
This is a hack-fix but if you Answer the call before dialing, that might
remove the progress message
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 12:47 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Please Advice SIP 183 progessl
Guys I tried the R,r options at the DIAL(SIP/,,rm) and also its sending
183
Any Advice
Dears
Kindly find my dial script below,I am trying to send the caller 180 ringing
but all tries were failed,
The caller always receive 183 session Progress
Even I add in the sip.conf
progressinband=never
exten = _X.,1,Wait(1)
exten = _X.,n,SetMusicOnHold(English)
exten = _X.,n,WaitMusicOnHold(2)
exten = _X.,n,NoOp(Return-)
-- add --
Exten = _X.,n,Answer()
-- end add --
exten = _X.,n,Dial(SIP/Gateway1/8430${EXTEN}|300|m)
;exten = _X.,n,SIPAddHeader(Alert-Info: Ring Answer)
exten = _X.,n,Goto(y-${DIALSTATUS},1) ; Jump based on status
(NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten = y-NOANSWER,1,SetMusicOnHold(busy)
exten = y-NOANSWER,n,WaitMusicOnHold(18) ; If busy, Playback Busy /
NOANSWER announce
exten = y-BUSY,1,SetMusicOnHold(busy)
exten = y-BUSY,n,WaitMusicOnHold(18) ; If busy, Playback Busy / busy
announce
exten = _y-.,1,Goto(y-NOANSWER,1) ; Treat anything else as no answer
exten = _X.,n,HangUp()
Please Advice
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:40 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] SIP 183 progessl
Sipaddheader(180 Ringing) might do the trick.
If you are compiling your own asterisk, you could change chan_sip.c to
replace 183 Session Progress with 180 Ringing (line 3950 in my source)
but that might break something else.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:23 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] SIP 183 progessl
Can you please tell me how to Custom SIP header
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Thursday, April 02, 2009 6:16 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Custom SIP header?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W.
Chehab
Sent: Thursday, April 02, 2009 10:02 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: Re: [asterisk-users] Xorcom and Doorbell
Dears
How can I send or force sending 180 Ringing instead of 183 back to the
caller ?or send both sequential if its impossible
I used progressinband=never but it did work .
Regards
*
No employee or agent is authorized to conclude any binding agreement on
behalf of Xplorium with another party by e-mail without express written
confirmation by an officer of Xplorium. Any views expressed by an individual
in this electronic message do not necessarily reflect views of Xplorium or
its subsidiaries and associates.
This electronic message and its attachments are solely addressed to the
addressee(s), and contain confidential
[asterisk-users] Early Media
Dears, - Anyone know how to play an early media as (background song) with no billing and when the call is connected the song will stop and the billing starts. Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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] Early Media
What I am meaning is . I want to start a music on hold and dial the number (009713045212) In the same time and when the call is connected the music will stop and I will talk to the called number Exten = 444,1,-- exten = 444,n,Dial(SIP/OutGoingGateway/009713045212|300|) is it feasible regards From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas Sent: Wednesday, March 25, 2009 3:34 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Early Media YMMV, but you might try this Exten = s,1,background(background_song) Exten = s,n,Answer() ;start billing _ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Khaled W. Chehab Sent: Wednesday, March 25, 2009 8:27 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: [asterisk-users] Early Media Dears, - Anyone know how to play an early media as (background song) with no billing and when the call is connected the song will stop and the billing starts. Regards _ * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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 Differences
Dears What's the major deference between Asterisk 1.6.0.6 and Asterisk 1.4.23 Regards Khaled Chehab NGN Eng. Untitled Operations Office - Lebanon Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: mailto:bs...@mg-tel.com kche...@xplorium.com MSN ID :khalidche...@hotmail.com Web Site: http://www.Xplorium.com * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * image001.jpg___ -- 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 Differences
Thanks,and kindly in which version of asterisk you advice to build a business PBX ? Regards -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus Darilion Sent: Thursday, March 05, 2009 12:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk Differences 1.6.0.6 - 1.4.23 -- 0.1.77.6 :-) http://svn.digium.com/view/asterisk/branches/1.6.0/CHANGES?revision=172635v iew=co klaus Khaled W. Chehab schrieb: Dears What's the major deference between Asterisk 1.6.0.6 and Asterisk 1.4.23 Regards** * Khaled Chehab* * NGN Eng.* Untitled * Operations Office - Lebanon* Office : +961 1 868686 ext 115 Mobile: +961 3 045212 E-mail: kche...@xplorium.com mailto:bs...@mg-tel.com MSN ID :khalidche...@hotmail.com Web Site: http://www.Xplorium.com * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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 * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ -- 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
