[asterisk-users] Which files to be copied
Hi List; I need to do upgrade for Asterisk and Zaptel, so which directories or files need to be copied to keep my configuration? Is it only the /etc directory or there is other directories? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Problem
I think you haven't capture the packet from the beginning of the call. You must capture the SIP packets. And the wireshark will recognise the packets as RTP. 木木 2007-11-16 发件人： Benjamin Jacob 发送时间： 2007-11-16 12:55:51 收件人： Asterisk Users Mailing List - Non-Commercial Discussion 抄送： 主题： Re: [asterisk-users] DTMF Problem for UDP tcpdump -nnXs 0 udp -i eth0 -w name.cap Btw, a pcap file (created on a linux server using tcpdump) capturing the RTP(udp) traffic opened up in wireshark, wireshark doesn't really format(or recognize) the packets as RTP, unlike the capture done live from a wireshark configured to capture RTP traffic. In the former, wireshark shows up everything as UDP and I have to do a lot of manual parsing to find out the type etc in the packets captured. Am I missing some config on wireshark here? TiA - Ben. ľľ wrote: You can use the tcpdump comand in linux. Like: tcpdump -i eth0 -s 0 -w name.cap And you can open the cap file useing wireshark that is a good 木木 2007-11-16 *发件人：* Doug *发送时间：* 2007-11-16 00:53:15 *收件人：* Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - No *抄送：* *主题：* Re: [asterisk-users] DTMF Problem At 06:42 11/15/2007, =?gb2312?B?xL7Evg==?= wrote: Hi， Could you capture the the UDP package How is this done? in all of your server, Asterisk A, Asterisk B, ser, Asterisk C. And you can find that server who lost the DTMF (RTP EVENT). -- Amy 2007-11-15 -- 发件人： Arun Kumar 发送时间： 2007-11-15 20:30:45 收件人： Asterisk Users Mailing List - Non-Commercial Discussion; SER Users 抄送： 主题： [asterisk-users] DTMF Problem Hi Here is my setup: USER -- PSTN - Asterisk A IAX2 Trunk Asterisk B - SER Asterisk C I'm not able to receive DTMF passed by USER on Asterisk C. All my asterisk boxs are configured with same DTMF type (auto) but no luck. Please help on this issue. Thanks, Arun ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dialogic
On Nov 15, 2007 11:01 AM, Steve Totaro wrote: [...] I may be wrong, wouldn't be the first time but I think you need to buy ABE to use Dialogic boards. If that is not correct, someone please correct me. thanks steve, I didn't know that they were supported. I looked in the compatibility list, none were listed : http://www.digium.com/en/supportcenter/documentation/viewdocs/ABE It is nice to know that a commercial version exists that supports dialogic cards : http://www.digium.com/en/products/software/abe.php?tab=overview -baji. -- ___ --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] r2 multiframe error - continue
Im using libs from astunicall-1.4.9-0.1.tar.gz at http://www.moythreads.com/astunicall/downloads/ (i have reinstalled asterisk, and libs from this package once again) No one can call me and i cant call out. Man from teleco still have teletransmision error.. No after starting asterisk im getting in full log something like this: [Nov 16 13:22:09] NOTICE[3787] chan_unicall.c: Unicall/17 event Detected [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 - C on [2/DETECTED/Seize ack /Seize ack] [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 R2 prot. err. [2/DETECTED/Seize ack /Seize ack] cause 32772 - Unexpected MF6 signal [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 1001 - [1/IDLE/Idle /Idle ] [Nov 16 13:22:09] NOTICE[3787] chan_unicall.c: Unicall/17 event Protocol failure [Nov 16 13:22:09] ERROR[3787] chan_unicall.c: Unicall/17 protocol error. Cause 32772 [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 Channel echo cancel [Nov 16 13:22:09] DEBUG[3787] chan_unicall.c: disabled echo cancellation on channel 17 What can i do?:) Regards Arkon ___ --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] make config update-rc.d
On Thu, Nov 15, 2007 at 06:47:04PM +0100, Philipp Kempgen wrote: On Debian the Asterisk Makefile does /usr/sbin/update-rc.d asterisk start 10 2 3 4 5 . stop 91 2 3 4 5 .; which results in a /etc/rc2.d/S10asterisk being written. I think S10 is too early. And it would also be simpler to use: update-rc.d asterisk defaults 10 91 Or, for better numbers: update-rc.d asterisk defaults 30 15 Note that on Debian the K scripts are in runlevels 0, 1 and 6. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changing audio message to text message
Hi all, I know Asterisk is able to send a waiting message (audio) to people trying to call a busy user agent using a queue. However, I'd like to change this audio message to a text message to be able to print it on screen on the other end. Is it possible to configure Asterisk to have text message sent ? Thanks, -- Anthony Chapellier - MBDSYS SARL 1, centre commercial de la Tour 93120 LA COURNEUVE FRANCE E-mail : [EMAIL PROTECTED] Tel : +33 (0) 143 11 09 14 ou +33 (0) 148 35 20 46 Fax : +33 (0) 148 37 79 28 http://www.mbdsys.com ___ --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] Help with Polycom 320
Is there anything in the CLI about the sip peer? Can you show the settings you have in sip.conf and the phone setting you entered? Bruce Reeves On Nov 16, 2007 9:00 AM, Jarga Jallow [EMAIL PROTECTED] wrote: I am having trouble configuring my Polycom 320 IP phone. When I dial an extension it seems like am calling from outside. Also on the phone menu it says not registered. Does anybody know how to fix this? Thanks in advance Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 www.2mcctv.com ___ --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 -- Bruce Reeves Nortex Networks ___ --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] r2 multiframe error - continue
Send all messages for the situation you have the problem.. It looks like to receive a C code really unexpected ... Luis A P Barbosa 2007/11/16, Jakub Syrek [EMAIL PROTECTED]: I was testing my system in local loop for protocolvariant mx,3,3(e1 cross cable between two spans). Here are results: testcall Loading protocol mfcr2 Thread for channel 0 MFC/R2 Chan 41: Call control(9) MFC/R2 Chan 41: Unblock MFC/R2 Chan 41: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 41: far_unblocking_expired MFC/R2 Chan 41: local_unblocking_expired Chan 41: -- Far end unblocked! :-) Chan 41: -- Far end unblocked! :-) Chan 41: -- Local end unblocked! :-) Chan 41: -- Local end unblocked! :-) Chan 41: Initiating call MFC/R2 Chan 41: Call control(1) MFC/R2 Chan 41: Make call MFC/R2 Chan 41: Creating a new call with CRN 32769 MFC/R2 Chan 41: 0001 - [1/DIALING /Seize /Idle ] Chan 41: -- Dialing on channel 0 Chan 41: -- Dialing on channel 0 MFC/R2 Chan 41: - 1101 [1/DIALING /Seize /Idle ] MFC/R2 Chan 41: 1 on - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 6 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 1 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 6 off [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: Calling party category 0x0 MFC/R2 Chan 41: 1 on - [2/DIALING /Group III /Category ] MFC/R2 Chan 41: - 5 on [2/DIALING /Group III /Category ] MFC/R2 Chan 41: 1 off - [2/DIALING /Group III /Category ] MFC/R2 Chan 41: - 5 off [2/DIALING /Group III /Category ] MFC/R2 Chan 41: 2 on - [2/DIALING /Group III /DNIS ] MFC/R2 Chan 41: - 1 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 2 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 off [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 3 on - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 3 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 off [2/DIALING /Group I /DNIS ] Main thread MFC/R2 Chan 41: - 3 on [2/DIALING /Group I /Silent ] MFC/R2 Chan 41: - 3 off [2/DIALING /Group I /Silent ] MFC/R2 Chan 41: 1 on - [2/PROCEED /Group II /Category ] Chan 41: -- Proceeding on channel 0 MFC/R2 Chan 41: - 1 on [2/PROCEED /Group II /Category ] MFC/R2 Chan 41: 1 off - [2/PROCEED /Group II /Category ] MFC/R2 Chan 41: - 1 off [2/PROCEED /Group II /Category ] Chan 41: -- Alerting on channel 0 Chan 41: -- Alerting on channel 0 MFC/R2 Chan 41: - 0101 [1/ALERTING/Await answer /Category ] Chan 41: -- Connected on channel 0 Chan 41: -- Connected on channel 0 Chan 41: -- '*0001*343*123*#' Main thread Main thread Main thread MFC/R2 Chan 41: - 1101 [1/CONNECTD/Answered /Category ] MFC/R2 Chan 41: Far end disconnected(cause=Normal Clearing [16]) - state 0x400 and asterisk log [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/25 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/26 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/27 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/28 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/29 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/30 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/31 event Local end unblocked [Nov 16 18:07:07] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 1001 [1/BLOCKED /Idle /Idle ] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 far_unblocking_expired [Nov 16 18:07:08] NOTICE[28848] chan_unicall.c: Unicall/10 event Far end unblocked [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 0001 [1/IDLE/Idle /Idle ] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 Detected [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 Creating a new call with CRN 32769 [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 1101 - [2/DETECTED/Seize ack /Seize ack] [Nov 16 18:07:08] NOTICE[28848] chan_unicall.c: Unicall/10 event Detected [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 1 on [2/DETECTED/Seize ack /Seize ack] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 6 on - [2/DETECTED/Group C /Category req ] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 1 off
Re: [asterisk-users] r2 multiframe error - continue
So, that means it is succeeded for mx protocolvariant. Now, just change the protocolvariant 'mx' to whatever fits your country, change only the country, please leave ANI,DNIS,OPTIONS . If it fails, I think a bug exists in your particular protocolvariant. Let me know the results. On Nov 16, 2007 11:11 AM, Jakub Syrek [EMAIL PROTECTED] wrote: I was testing my system in local loop for protocolvariant mx,3,3(e1 cross cable between two spans). Here are results: testcall Loading protocol mfcr2 Thread for channel 0 MFC/R2 Chan 41: Call control(9) MFC/R2 Chan 41: Unblock MFC/R2 Chan 41: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 41: far_unblocking_expired MFC/R2 Chan 41: local_unblocking_expired Chan 41: -- Far end unblocked! :-) Chan 41: -- Far end unblocked! :-) Chan 41: -- Local end unblocked! :-) Chan 41: -- Local end unblocked! :-) Chan 41: Initiating call MFC/R2 Chan 41: Call control(1) MFC/R2 Chan 41: Make call MFC/R2 Chan 41: Creating a new call with CRN 32769 MFC/R2 Chan 41: 0001 - [1/DIALING /Seize /Idle ] Chan 41: -- Dialing on channel 0 Chan 41: -- Dialing on channel 0 MFC/R2 Chan 41: - 1101 [1/DIALING /Seize /Idle ] MFC/R2 Chan 41: 1 on - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 6 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 1 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 6 off [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: Calling party category 0x0 MFC/R2 Chan 41: 1 on - [2/DIALING /Group III /Category ] MFC/R2 Chan 41: - 5 on [2/DIALING /Group III /Category ] MFC/R2 Chan 41: 1 off - [2/DIALING /Group III /Category ] MFC/R2 Chan 41: - 5 off [2/DIALING /Group III /Category ] MFC/R2 Chan 41: 2 on - [2/DIALING /Group III /DNIS ] MFC/R2 Chan 41: - 1 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 2 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 off [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 3 on - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 3 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 off [2/DIALING /Group I /DNIS ] Main thread MFC/R2 Chan 41: - 3 on [2/DIALING /Group I /Silent ] MFC/R2 Chan 41: - 3 off [2/DIALING /Group I /Silent ] MFC/R2 Chan 41: 1 on - [2/PROCEED /Group II /Category ] Chan 41: -- Proceeding on channel 0 MFC/R2 Chan 41: - 1 on [2/PROCEED /Group II /Category ] MFC/R2 Chan 41: 1 off - [2/PROCEED /Group II /Category ] MFC/R2 Chan 41: - 1 off [2/PROCEED /Group II /Category ] Chan 41: -- Alerting on channel 0 Chan 41: -- Alerting on channel 0 MFC/R2 Chan 41: - 0101 [1/ALERTING/Await answer /Category ] Chan 41: -- Connected on channel 0 Chan 41: -- Connected on channel 0 Chan 41: -- '*0001*343*123*#' Main thread Main thread Main thread MFC/R2 Chan 41: - 1101 [1/CONNECTD/Answered /Category ] MFC/R2 Chan 41: Far end disconnected(cause=Normal Clearing [16]) - state 0x400 and asterisk log [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/25 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/26 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/27 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/28 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/29 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/30 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/31 event Local end unblocked [Nov 16 18:07:07] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 1001 [1/BLOCKED /Idle /Idle ] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 far_unblocking_expired [Nov 16 18:07:08] NOTICE[28848] chan_unicall.c: Unicall/10 event Far end unblocked [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 0001 [1/IDLE/Idle /Idle ] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 Detected [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 Creating a new call with CRN 32769 [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 1101 - [2/DETECTED/Seize ack /Seize ack] [Nov 16 18:07:08] NOTICE[28848] chan_unicall.c: Unicall/10 event Detected [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 1 on [2/DETECTED/Seize ack /Seize ack] [Nov 16 18:07:08] WARNING[28848]
Re: [asterisk-users] r2 multiframe error - continue
I was testing my system in local loop for protocolvariant mx,3,3(e1 cross cable between two spans). Here are results: testcall Loading protocol mfcr2 Thread for channel 0 MFC/R2 Chan 41: Call control(9) MFC/R2 Chan 41: Unblock MFC/R2 Chan 41: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 41: far_unblocking_expired MFC/R2 Chan 41: local_unblocking_expired Chan 41: -- Far end unblocked! :-) Chan 41: -- Far end unblocked! :-) Chan 41: -- Local end unblocked! :-) Chan 41: -- Local end unblocked! :-) Chan 41: Initiating call MFC/R2 Chan 41: Call control(1) MFC/R2 Chan 41: Make call MFC/R2 Chan 41: Creating a new call with CRN 32769 MFC/R2 Chan 41: 0001 - [1/DIALING /Seize /Idle ] Chan 41: -- Dialing on channel 0 Chan 41: -- Dialing on channel 0 MFC/R2 Chan 41: - 1101 [1/DIALING /Seize /Idle ] MFC/R2 Chan 41: 1 on - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 6 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 1 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 6 off [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: Calling party category 0x0 MFC/R2 Chan 41: 1 on - [2/DIALING /Group III /Category ] MFC/R2 Chan 41: - 5 on [2/DIALING /Group III /Category ] MFC/R2 Chan 41: 1 off - [2/DIALING /Group III /Category ] MFC/R2 Chan 41: - 5 off [2/DIALING /Group III /Category ] MFC/R2 Chan 41: 2 on - [2/DIALING /Group III /DNIS ] MFC/R2 Chan 41: - 1 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 2 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 off [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 3 on - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 3 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 off [2/DIALING /Group I /DNIS ] Main thread MFC/R2 Chan 41: - 3 on [2/DIALING /Group I /Silent ] MFC/R2 Chan 41: - 3 off [2/DIALING /Group I /Silent ] MFC/R2 Chan 41: 1 on - [2/PROCEED /Group II /Category ] Chan 41: -- Proceeding on channel 0 MFC/R2 Chan 41: - 1 on [2/PROCEED /Group II /Category ] MFC/R2 Chan 41: 1 off - [2/PROCEED /Group II /Category ] MFC/R2 Chan 41: - 1 off [2/PROCEED /Group II /Category ] Chan 41: -- Alerting on channel 0 Chan 41: -- Alerting on channel 0 MFC/R2 Chan 41: - 0101 [1/ALERTING/Await answer /Category ] Chan 41: -- Connected on channel 0 Chan 41: -- Connected on channel 0 Chan 41: -- '*0001*343*123*#' Main thread Main thread Main thread MFC/R2 Chan 41: - 1101 [1/CONNECTD/Answered /Category ] MFC/R2 Chan 41: Far end disconnected(cause=Normal Clearing [16]) - state 0x400 and asterisk log [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/25 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/26 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/27 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/28 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/29 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/30 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/31 event Local end unblocked [Nov 16 18:07:07] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 1001 [1/BLOCKED /Idle /Idle ] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 far_unblocking_expired [Nov 16 18:07:08] NOTICE[28848] chan_unicall.c: Unicall/10 event Far end unblocked [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 0001 [1/IDLE/Idle /Idle ] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 Detected [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 Creating a new call with CRN 32769 [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 1101 - [2/DETECTED/Seize ack /Seize ack] [Nov 16 18:07:08] NOTICE[28848] chan_unicall.c: Unicall/10 event Detected [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 1 on [2/DETECTED/Seize ack /Seize ack] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 6 on - [2/DETECTED/Group C /Category req ] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 1 off [2/DETECTED/Group C /Category req ] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 6 off - [2/DETECTED/Group C /Category req ] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 1 on [2/DETECTED/Group C
[asterisk-users] channels to destroy
Hello, In a couple of Asterisks, after type sip show channels we have a lot of these: IP_PEER dst_number something00102/00103 unkn No (d) Rx: BYE IP_PEER dst_number2 something2 00102/00103 unkn No (d) Rx: BYE We are using ASterisk 1.2.x When I say a lot I mean more than 180, more than 230, etc. Is it normal? How we can remove it? Thank you very much, -- Carles Pina i EstanyGPG id: 0x8CBDAE64 http://pinux.info Manresa - Barcelona ___ --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] 1.4 SIP Jitter Buffer
I have asterisk 1.4.14 in 3 of my 8 servers for 3 weeks on productions systems, but i had problem with adapative jitter buffer, when i active it there are no sound. Regards On Nov 6, 2007 9:16 PM, Gregory Boehnlein [EMAIL PROTECTED] wrote: Are you running the SIP Jitter Buffer? -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Luc Moreira Sent: Monday, November 05, 2007 10:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 SIP Jitter Buffer Gregory We have many Asterisk 1.4.13 in production solid like a rock. Couples examples: a) Asterisk 1.4.13 + Unicall + 2 E1 MFCR2 Digium + Legacy PBX 60+ Extentions / IVR / 10~30 concorrent calls b) Asterisk 1.4.11 + 1 E1 ISDN PRI Digium 50+ Extentions / IVR / 5 Queues / ~2000 call/day c) Asterisk 1.4.13 + 4 E1 ISDN Digium (working in progress) CallCenter / 150 PAs / 15 Queues / expected 8000 calls/day -- Luc Gregory Boehnlein escreveu: Hello, I'm running into a few situations on lossy network links where a SIP jitter buffer w/ some PLC would be helpful. My main TDM gateways are running 1.2 (which is solid, stable, reliable and very very very well behaved when you know it's limitations), but I'm considering upgrading them before the end of the year to 1.4. Two of the main reasons that I would do this are Variable Length DTMF and SIP Jitter Buffering. I would be very interested in hearing from anyone that is actually running 1.4 in a PRODUCTION environment, gatewaying SIP to TDM using Digium cards. To me, production means being able to have 3-4 PRI circuits maxed out for 12+ hours a day and 7+ call setups / second. Anything less than that is not really going to be an accurate comparison to what I have running. Anyone have any feedback about this combination? ___ --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 -- This message has been scanned for viruses and dangerous content by N2Net Mailshield, and is believed to be clean. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with AGI Script
On Wed, 14 Nov 2007, Moises Silva wrote: what does agi debug says? what if you run the script from the command line and you fake the asterisk input? agi debug (and syslog()) are the AGI developer's best friends :) It shows the dialog between Asterisk and your AGI. Even if your AGI fails to exist or execute, you will still see Asterisk setting up the AGI environment for you, just none of the subsequent dialog. I use the following script to fake the AGI environment: # the standard AGI environment variables echo agi_accountcode: echo agi_callerid: 1234567890 echo agi_calleridname: sedwards echo agi_callingani2: 0 echo agi_callingpres: 0 echo agi_callingtns: 0 echo agi_callington: 0 echo agi_channel: SIP/201-09456478 echo agi_context: newline echo agi_dnid: * echo agi_enhanced: 0.0 echo agi_extension: * echo agi_language: en echo agi_priority: 3 echo agi_rdnis: unknown echo agi_request: block-ani echo agi_type: SIP echo agi_uniqueid: 1195070681.28 echo # cruft specific to my AGI # result for AGI command SET PRIORITY echo 200 result=0 # result for AGI command GET VARIABLE ANI echo 200 result=1 (1234567890) # result for AGI command GET VARIABLE CALLINGANI2 echo 200 result=1 (0) # result for AGI command GET VARIABLE DATABASE-SERVER echo 200 result=1 (pdb) # result for AGI command SET PRIORITY echo 200 result=0 # (end of agi-environment.sh) This can be used as: chmod +x agi-environment.sh ./agi-environment.sh | block-ani --debug I use getopt_long() to parse the command line options and syslog() to stash stuff to analyze after the fact. You can even use gdb (if appropriate for your language) by saving the output of the script and supplying it in the r command as: ./agi-environment.sh agi-environment gdb block-ani (gdb) r --debug agi-environment Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which files to be copied
If you want to be completely safe, copy everything from /var/lib/asterisk, /var/spool/asterisk, and /var/log/asterisk as well. On Fri, 16 Nov 2007, bilal ghayyad wrote: Hi List; I need to do upgrade for Asterisk and Zaptel, so which directories or files need to be copied to keep my configuration? Is it only the /etc directory or there is other directories? Regards Bilal Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs ___ --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 -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files
2007/11/15, Greg Oliver [EMAIL PROTECTED]: On Thu, 2007-11-15 at 07:31 +0100, Olivier wrote: 2007/11/14, Greg Oliver [EMAIL PROTECTED]: On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Softkeys running both SCCP and SIP firmware are both sent through the protocols themselves. How ? In SIP mode, is it using RegEvents (rfc3680) ? regards Cisco using RFCs - lol - I wish... Without softkey configuration files, I've heard you cannot translate menus when connecting a Cisco SIP phone to any non-Cisco SIP server. -Greg ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reload command
Well, during reload all modules, including channel technologies will call reload(), so it is hard to tell if it is a bug or just that the channel driver is not ready to make calls when you ask to make a call due to the reload(). More information is needed. Does this happen with any technology, do you enabled ALL logs? if so, can you find any relevant ERROR, WARNING or something like that when the call fails? - Moy On Nov 15, 2007 9:22 PM, Jerry Geis [EMAIL PROTECTED] wrote: What do you mean by reload. Please be specific. Reload or restart. Reload which module? or did you mean restart Asterisk? On Nov 15, 2007 7:12 AM, Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: / All, // // I have noticed that placing a call in the outgoing spool during a reload // the call may fail. Try the call again after the reload is done and it will // complete. // // This seems like a bug. During a reload calls should be suspended or // something? // // Thoughts? // // Jerry/ I'm talking about the command: /usr/sbin/asterisk -rx reload Jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --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] Dialogic
thats correct, but i want to talk/know if someone else is using this cards withs asterisk. thanks nico On Thu, 15 Nov 2007, Steve Totaro wrote: [EMAIL PROTECTED] wrote: Hi, Is anybody there who can give me information which things are supported in Asterisk on Dialogic E1-Cards? Does anybody use a Dialogic card in Asterisk? (Not the DIVA, i know eicon is using some ISDN-Stack which is working in Asterisk) I mean the really old Dialogic cards. thanks Nico I may be wrong, wouldn't be the first time but I think you need to buy ABE to use Dialogic boards. If that is not correct, someone please correct me. Thanks, Steve Totaro 888.777.1888 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dumb AGI question
On Fri, 16 Nov 2007, David Ruggles wrote: I did some simple AGI programming several months ago. I have a need to use one of those old programs and I'm having a stupid problem. I can't get output to display on the console. I'm sending it to stderr and I've got verbosity set to 10. I know I had it working before so I'm guessing I just forgot some piece of key information. Any suggestions? Jumping up on my syslog soapbox... Check out man syslog. If it fits your needs, logging to syslog is trivial. For example: syslog(LOG_ERR, The value of foo is %s, bar); If not, how about: ) Did you read all of the AGI environment from stdin? ) Why not use the agi command verbose? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-300 ___ --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] r2 multiframe error - continue
Im from Poland and there is no pl option, what should i chose? Arkon - Original Message - From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 16, 2007 7:01 PM Subject: Re: [asterisk-users] r2 multiframe error - continue So, that means it is succeeded for mx protocolvariant. Now, just change the protocolvariant 'mx' to whatever fits your country, change only the country, please leave ANI,DNIS,OPTIONS . If it fails, I think a bug exists in your particular protocolvariant. Let me know the results. On Nov 16, 2007 11:11 AM, Jakub Syrek [EMAIL PROTECTED] wrote: I was testing my system in local loop for protocolvariant mx,3,3(e1 cross cable between two spans). Here are results: testcall Loading protocol mfcr2 Thread for channel 0 MFC/R2 Chan 41: Call control(9) MFC/R2 Chan 41: Unblock MFC/R2 Chan 41: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 41: far_unblocking_expired MFC/R2 Chan 41: local_unblocking_expired Chan 41: -- Far end unblocked! :-) Chan 41: -- Far end unblocked! :-) Chan 41: -- Local end unblocked! :-) Chan 41: -- Local end unblocked! :-) Chan 41: Initiating call MFC/R2 Chan 41: Call control(1) MFC/R2 Chan 41: Make call MFC/R2 Chan 41: Creating a new call with CRN 32769 MFC/R2 Chan 41: 0001 - [1/DIALING /Seize /Idle ] Chan 41: -- Dialing on channel 0 Chan 41: -- Dialing on channel 0 MFC/R2 Chan 41: - 1101 [1/DIALING /Seize /Idle ] MFC/R2 Chan 41: 1 on - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 6 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 1 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 6 off [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: Calling party category 0x0 MFC/R2 Chan 41: 1 on - [2/DIALING /Group III /Category ] MFC/R2 Chan 41: - 5 on [2/DIALING /Group III /Category ] MFC/R2 Chan 41: 1 off - [2/DIALING /Group III /Category ] MFC/R2 Chan 41: - 5 off [2/DIALING /Group III /Category ] MFC/R2 Chan 41: 2 on - [2/DIALING /Group III /DNIS ] MFC/R2 Chan 41: - 1 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 2 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 off [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 3 on - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 3 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 off [2/DIALING /Group I /DNIS ] Main thread MFC/R2 Chan 41: - 3 on [2/DIALING /Group I /Silent ] MFC/R2 Chan 41: - 3 off [2/DIALING /Group I /Silent ] MFC/R2 Chan 41: 1 on - [2/PROCEED /Group II /Category ] Chan 41: -- Proceeding on channel 0 MFC/R2 Chan 41: - 1 on [2/PROCEED /Group II /Category ] MFC/R2 Chan 41: 1 off - [2/PROCEED /Group II /Category ] MFC/R2 Chan 41: - 1 off [2/PROCEED /Group II /Category ] Chan 41: -- Alerting on channel 0 Chan 41: -- Alerting on channel 0 MFC/R2 Chan 41: - 0101 [1/ALERTING/Await answer /Category ] Chan 41: -- Connected on channel 0 Chan 41: -- Connected on channel 0 Chan 41: -- '*0001*343*123*#' Main thread Main thread Main thread MFC/R2 Chan 41: - 1101 [1/CONNECTD/Answered /Category ] MFC/R2 Chan 41: Far end disconnected(cause=Normal Clearing [16]) - state 0x400 and asterisk log [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/25 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/26 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/27 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/28 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/29 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/30 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/31 event Local end unblocked [Nov 16 18:07:07] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 1001 [1/BLOCKED /Idle /Idle ] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 far_unblocking_expired [Nov 16 18:07:08] NOTICE[28848] chan_unicall.c: Unicall/10 event Far end unblocked [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 0001 [1/IDLE/Idle /Idle ] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 Detected [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 Creating a new call with CRN 32769 [Nov 16
Re: [asterisk-users] Which files to be copied
The other directories were mentioned to backup your voicemail messages, voicemail greetings, custom sound files, and historical logs. If you don't need any of that, it's up to you. Yes, the configuration files are solely in /etc/asterisk/ typically, but most people will want to keep their voicemail greetings and messages and such when they upgrade. Moj bilal ghayyad wrote: Dear Alex; Thanks for your help :) - But here there will be a question: new installation will write for which directories? In case I copied the var/lib/asterisk and var/spool/asterisk then that means I will use these data incase I need to come back for my original version after the upgrade, correct? About configuration files, they are only in /etc/ and /etc/asterisk? Regards Bilal If you want to be completely safe, copy everything from /var/lib/asterisk, /var/spool/asterisk, and /var/log/asterisk as well. On Fri, 16 Nov 2007, bilal ghayyad wrote: Hi List; I need to do upgrade for Asterisk and Zaptel, so which directories or files need to be copied to keep my configuration? Is it only the /etc directory or there is other directories? Regards Bilal Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See how. http://overview.mail.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dumb AGI question
Thank you sir! Am using verbose command now which works and I will take a look at the syslog. Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Edwards Sent: Friday, November 16, 2007 2:07 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Dumb AGI question On Fri, 16 Nov 2007, David Ruggles wrote: I did some simple AGI programming several months ago. I have a need to use one of those old programs and I'm having a stupid problem. I can't get output to display on the console. I'm sending it to stderr and I've got verbosity set to 10. I know I had it working before so I'm guessing I just forgot some piece of key information. Any suggestions? Jumping up on my syslog soapbox... Check out man syslog. If it fits your needs, logging to syslog is trivial. For example: syslog(LOG_ERR, The value of foo is %s, bar); If not, how about: ) Did you read all of the AGI environment from stdin? ) Why not use the agi command verbose? Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-300 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files
On Thu, 2007-11-15 at 07:31 +0100, Olivier wrote: 2007/11/14, Greg Oliver [EMAIL PROTECTED]: On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Softkeys running both SCCP and SIP firmware are both sent through the protocols themselves. How ? In SIP mode, is it using RegEvents (rfc3680) ? regards Cisco using RFCs - lol - I wish... -Greg ___ --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 1.4 with LDAP
Hi friends. How do I can use Asterisk 1.4 with LDAP? I need it because the system must use just one password for each user for everything. A lot of thanks. -- Linux User Registered #232544 Jabber : [EMAIL PROTECTED] Ekiga : [EMAIL PROTECTED] ICQ : 337889406 GnuPG-key : www.keyserver.net --- dum loquimur, fugerit invida aetas: carpe diem, quam minimum credula postero. ___ --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] make config update-rc.d
On Thursday 15 November 2007 11:47:04 Philipp Kempgen wrote: On Debian the Asterisk Makefile does /usr/sbin/update-rc.d asterisk start 10 2 3 4 5 . stop 91 2 3 4 5 .; which results in a /etc/rc2.d/S10asterisk being written. I think S10 is too early. bind9 : S15 mysql : S19 zaptel: S20 ntp : S23 What bothers me most is that mysql is not up when asterisk starts. That's a bad thing if there are #execs in your config files and if the scripts rely on mysql. So what about S50 or S95? Okay, changed. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Which files to be copied
Dear Alex; Thanks for your help :) - But here there will be a question: new installation will write for which directories? In case I copied the var/lib/asterisk and var/spool/asterisk then that means I will use these data incase I need to come back for my original version after the upgrade, correct? About configuration files, they are only in /etc/ and /etc/asterisk? Regards Bilal If you want to be completely safe, copy everything from /var/lib/asterisk, /var/spool/asterisk, and /var/log/asterisk as well. On Fri, 16 Nov 2007, bilal ghayyad wrote: Hi List; I need to do upgrade for Asterisk and Zaptel, so which directories or files need to be copied to keep my configuration? Is it only the /etc directory or there is other directories? Regards Bilal Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See how. http://overview.mail.yahoo.com/ ___ --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 Program Closes
At 11:33 11/15/2007, Ryan M. Colbert wrote: Periodically, maybe once or twice every few weeks, we see our instance of Asterisk 1.4.7 just close out without warning and we have to reload the module. Were running CentOS. Has anyone else seen this before? Core show version: Asterisk 1.4.7 built by root @ XX on a i686 running Linux on 2007-07-11 00:21:57 UTC Uname a: Linux XX 2.6.9-55.0.2.EL #1 Tue Jun 26 14:08:18 EDT 2007 i686 athlon i386 GNU/Linux There probably have been a few bugs fixed since 1.4.7. Latest version is 1.4.13: http://www.asterisk.org/downloads Perhaps running a Cron job to reboot in the middle of the night would solve some of these problems: crontab -e 0 4 * * * /sbin/shutdown -r now ___ --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] Change the Voice promps in asterisk 1.4
Hello all, Which is the best way to change the default Voice promps in asteriosk 1.4from english to french? And if I would like to add a new Voice promp set, how is the way to do? Thanks in advance. VoipCrazy ___ --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 Program Closes
Periodically, maybe once or twice every few weeks, we see our instance of Asterisk 1.4.7 just close out without warning and we have to reload the module. We're running CentOS. Has anyone else seen this before? Core show version: Asterisk 1.4.7 built by root @ XX on a i686 running Linux on 2007-07-11 00:21:57 UTC Uname -a: Linux XX 2.6.9-55.0.2.EL #1 Tue Jun 26 14:08:18 EDT 2007 i686 athlon i386 GNU/Linux Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ ___ --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] Music on Hold -- Error
We use Asterisk 1.4.7 on CentOS with Bandwidth.com as our provider and Polycom 330's for endpoints. When one of our end points places a call on hold we get the following in CLI. There is no music on hold provided for the caller. The SIP.CONF entry for our connection to bandwithd.com specifies disallow=all and allow=ulaw. Should there be a similar setting on the user.conf entries? An interesting note is the IP noted in the CLI message below is neither Bandwidth.com nor the end point. Thanks for any help!! CLI Message: [Nov 15 13:21:38] WARNING[11327]: channel.c:2964 set_format: Unable to find a codec translation path from ulaw to unknown [Nov 15 13:21:38] WARNING[11327]: res_musiconhold.c:702 moh_alloc: Unable to set channel 'SIP/4.68.250.148-08d1e7e0' to format 'unknown' Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Problem
for UDP tcpdump -nnXs 0 udp -i eth0 -w name.cap Btw, a pcap file (created on a linux server using tcpdump) capturing the RTP(udp) traffic opened up in wireshark, wireshark doesn't really format(or recognize) the packets as RTP, unlike the capture done live from a wireshark configured to capture RTP traffic. In the former, wireshark shows up everything as UDP and I have to do a lot of manual parsing to find out the type etc in the packets captured. Am I missing some config on wireshark here? TiA - Ben. ľľ wrote: You can use the tcpdump comand in linux. Like: tcpdump -i eth0 -s 0 -w name.cap And you can open the cap file useing wireshark that is a good 木木 2007-11-16 *发件人：* Doug *发送时间：* 2007-11-16 00:53:15 *收件人：* Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - No *抄送：* *主题：* Re: [asterisk-users] DTMF Problem At 06:42 11/15/2007, =?gb2312?B?xL7Evg==?= wrote: Hi， Could you capture the the UDP package How is this done? in all of your server, Asterisk A, Asterisk B, ser, Asterisk C. And you can find that server who lost the DTMF (RTP EVENT). -- Amy 2007-11-15 -- 发件人： Arun Kumar 发送时间： 2007-11-15 20:30:45 收件人： Asterisk Users Mailing List - Non-Commercial Discussion; SER Users 抄送： 主题： [asterisk-users] DTMF Problem Hi Here is my setup: USER -- PSTN - Asterisk A IAX2 Trunk Asterisk B - SER Asterisk C I'm not able to receive DTMF passed by USER on Asterisk C. All my asterisk boxs are configured with same DTMF type (auto) but no luck. Please help on this issue. Thanks, Arun ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] Two B-Channel Transfer (2BCT/TBCT) Trobule on DMS100 PRI
On Thu, 2007-11-15 at 13:51 -0500, Jacob Lefkowitz wrote: I have not been able to get two B-channel transfer to work on DMS100 PRI. I consistently get the following errors: I've successfully tested 2BCT on 5E switches, and it seems works to great. Just remember to set facilityenable=yes in zapata.conf. I'm not sure if my data point really helps you, as I was using a different type of switch, but I've heard from good authority that it's working on DMS100 as well. Please let us know if/when you find the answer... in the meantime, maybe someone else out there knows something more specific about the error messages you reported. -- Jared Smith Community Relations Manager Digium, Inc. ___ --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] Help with Polycom 320
I am having trouble configuring my Polycom 320 IP phone. When I dial an extension it seems like am calling from outside. Also on the phone menu it says not registered. Does anybody know how to fix this? Thanks in advance Jarga Jallow Technical Support Engineer 2985 S. Hwy. 360 Grand Praire, Texas 75052 Direct: 972-206-1212 ext# 29 Mobile: 214-669-9046 Fax:972-999-4113 Toll Free: 1-877-801-5511 ext 34 Toll Free: 1-877-926-2288 http://www.2mcctv.com/ www.2mcctv.com http://www.2mcctv.com/ image002.jpgimage003.gifimage004.gif___ --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] reload command
I'm talking about the command: /usr/sbin/asterisk -rx reload Jerry I am not sure why one would need reload these days unless you are making changes to the code in the modules. My guess is that you are doing this after a change to extensions.conf. If this is the case, please try /usr/sbin/asterisk -rx extensions reload or if you need to reload a specific module /usr/sbin/asterisk -rx module reload x Of course if you have a sip call in progress and you call reload on chan_sip, expect that the connection will be broken. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files
Sorry forgot the images: http://picasaweb.google.com/ranciso/AsteriskImagesCiscoPhones From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Anciso, Roy Sent: Thursday, November 15, 2007 3:47 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files The softkeys translate fine. Things like redial, new call, call forward, transfer, conference, hold, end call, do not disturb (for DND you have to go through a few more menus) line selection, services. I'll try attaching a screenshot of the softphone I have setup. I've setup the services button so you can browse the local extension directory (based on the sip.conf file) and I also setup a script to generate system speeds dials for all the phones. It also alphabetizes them automatically. I'm hoping to use a nonstandard template to make things like DND a bit more accessible. I just received the 7941 7911g phones from our Cisco rep I'm working on loading the SIP image on those. Oh the other thing I created is a script for auto generation of your SEPmac.cnf.xml file for each phone. You just enter in the mac address the sip extension, password, display name and phone label and the xml file is automatically generated. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Thursday, November 15, 2007 12:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files 2007/11/15, Greg Oliver [EMAIL PROTECTED]: On Thu, 2007-11-15 at 07:31 +0100, Olivier wrote: 2007/11/14, Greg Oliver [EMAIL PROTECTED]: On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Softkeys running both SCCP and SIP firmware are both sent through the protocols themselves. How ? In SIP mode, is it using RegEvents (rfc3680) ? regards Cisco using RFCs - lol - I wish... Without softkey configuration files, I've heard you cannot translate menus when connecting a Cisco SIP phone to any non-Cisco SIP server. -Greg ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] pip tones in Monitor or MixMonitor
I guess I didn't try this because Playback(beep) seems to me to playback the beep once and not repeat ever 15 seconds as is needed for the pip tones. Is this not true? exten = _X.,1,Playback(beep) exten = _X.,2,MixMonitor. If you are starting the recording using some DTMF code sequence described in features.conf make sure you use caller, callee or both value to play sound to correct line end. Mindaugas Kezys http://www.kolmisoft.com MOR - Advanced Billing for Asterisk PBX -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Plack Sent: Wednesday, November 14, 2007 11:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] pip tones in Monitor or MixMonitor Is there a way to enable the pip tones (beep) indicating that a call is being recorded? I know that ChanSpy does beep (unless q option is chosen) once, but not quite the same. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OT - best policy for logs
On Nov 15, 2007 12:55 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote: In my experience, it's easier to combine them all into one syslog server, and then utilize tools to filter them apart when necessary, since there are more tools to do that than to *combine* them when that is necessary, which it often is. Agreed - I have all of my servers send their syslogs to /var/log/messages on one central logging server. If you want to examine a device-specific log, just use tail + grep. That said, any system logger worth it's salt will make it extremely easy to have device-specific log files if you prefer. -erik ___ --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] Lists dead?
On Nov 15, 2007 12:32 PM, Philipp Kempgen wrote: Last message received at 2007-11-14 18:02:04 GMT not the case boss, there has been steady traffic for the past 6 hours, I estimate around 20 msgs, mostly replies. check your incoming server. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Problem with AGI Script
Steve Edwards wrote: On Thu, 15 Nov 2007, Benjamin Jacob wrote: well.. if nothings working.. try putting in debug lines urself in the code.. say use system calls to write some debugging data into some temporary file in ur perl code. I'm a big fan of syslog(LOG_ERR, I expected %d, but I got %d, foo, bar); to write a message to the system log. A single statement and no temporary files to clean up. Syslog has lots of features -- check out the man page. thats definitely better.. EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --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] reload command
What do you mean by reload. Please be specific. Reload or restart. Reload which module? or did you mean restart Asterisk? On Nov 15, 2007 7:12 AM, Jerry Geis geisj at pagestation.com http://lists.digium.com/mailman/listinfo/asterisk-users wrote: / All, // // I have noticed that placing a call in the outgoing spool during a reload // the call may fail. Try the call again after the reload is done and it will // complete. // // This seems like a bug. During a reload calls should be suspended or // something? // // Thoughts? // // Jerry/ I'm talking about the command: /usr/sbin/asterisk -rx reload Jerry ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] r2 multiframe error - continue
Wll, I think you should have started all this thread by mentioning that. May be libmfcr2 do not support R2 variant in poland. For you, the quick solution might be just ask your E1 in ISDN-PRI. If you really want to stay with R2 or you have no choice, we can arrange a meeting to start figuring out how to support poland R2 variant or a work-around for it. However I will not have any time before this wednesday. Have a great weekend. - Moy On Nov 16, 2007 1:07 PM, Jakub Syrek [EMAIL PROTECTED] wrote: Im from Poland and there is no pl option, what should i chose? Arkon - Original Message - From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 16, 2007 7:01 PM Subject: Re: [asterisk-users] r2 multiframe error - continue So, that means it is succeeded for mx protocolvariant. Now, just change the protocolvariant 'mx' to whatever fits your country, change only the country, please leave ANI,DNIS,OPTIONS . If it fails, I think a bug exists in your particular protocolvariant. Let me know the results. On Nov 16, 2007 11:11 AM, Jakub Syrek [EMAIL PROTECTED] wrote: I was testing my system in local loop for protocolvariant mx,3,3(e1 cross cable between two spans). Here are results: testcall Loading protocol mfcr2 Thread for channel 0 MFC/R2 Chan 41: Call control(9) MFC/R2 Chan 41: Unblock MFC/R2 Chan 41: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 41: far_unblocking_expired MFC/R2 Chan 41: local_unblocking_expired Chan 41: -- Far end unblocked! :-) Chan 41: -- Far end unblocked! :-) Chan 41: -- Local end unblocked! :-) Chan 41: -- Local end unblocked! :-) Chan 41: Initiating call MFC/R2 Chan 41: Call control(1) MFC/R2 Chan 41: Make call MFC/R2 Chan 41: Creating a new call with CRN 32769 MFC/R2 Chan 41: 0001 - [1/DIALING /Seize /Idle ] Chan 41: -- Dialing on channel 0 Chan 41: -- Dialing on channel 0 MFC/R2 Chan 41: - 1101 [1/DIALING /Seize /Idle ] MFC/R2 Chan 41: 1 on - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 6 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 1 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 6 off [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: Calling party category 0x0 MFC/R2 Chan 41: 1 on - [2/DIALING /Group III /Category ] MFC/R2 Chan 41: - 5 on [2/DIALING /Group III /Category ] MFC/R2 Chan 41: 1 off - [2/DIALING /Group III /Category ] MFC/R2 Chan 41: - 5 off [2/DIALING /Group III /Category ] MFC/R2 Chan 41: 2 on - [2/DIALING /Group III /DNIS ] MFC/R2 Chan 41: - 1 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 2 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 off [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 3 on - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 3 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 off [2/DIALING /Group I /DNIS ] Main thread MFC/R2 Chan 41: - 3 on [2/DIALING /Group I /Silent ] MFC/R2 Chan 41: - 3 off [2/DIALING /Group I /Silent ] MFC/R2 Chan 41: 1 on - [2/PROCEED /Group II /Category ] Chan 41: -- Proceeding on channel 0 MFC/R2 Chan 41: - 1 on [2/PROCEED /Group II /Category ] MFC/R2 Chan 41: 1 off - [2/PROCEED /Group II /Category ] MFC/R2 Chan 41: - 1 off [2/PROCEED /Group II /Category ] Chan 41: -- Alerting on channel 0 Chan 41: -- Alerting on channel 0 MFC/R2 Chan 41: - 0101 [1/ALERTING/Await answer /Category ] Chan 41: -- Connected on channel 0 Chan 41: -- Connected on channel 0 Chan 41: -- '*0001*343*123*#' Main thread Main thread Main thread MFC/R2 Chan 41: - 1101 [1/CONNECTD/Answered /Category ] MFC/R2 Chan 41: Far end disconnected(cause=Normal Clearing [16]) - state 0x400 and asterisk log [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/25 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/26 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/27 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/28 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/29 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/30 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/31 event Local end unblocked [Nov 16
[asterisk-users] Help on strange problem...
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hey all, I'm having problems with calls dropping after 15 - 20 seconds from a particular provider. The are using a NexTone gateway. Here are the details: Successful call: INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone INVITE (G711U) cseq 2 From NexTone 100 Trying cseq 2 From Asterisk 200 OK cseq 2 From Asterisk ACK cseq 2 From NexTone 200 OK (711U) cseq 1 From Asterisk ACK cseq 1 From NexTone Call continues until one side hangs up... Failed Call: Call audio is fine and all seems well but after 15 to 20 sec the call drops... INVITE cseq 1 From NexTone 100 Trying cseq 1 From Asterisk 100 Trying cseq 1 From Asterisk 200 OK (G711U) cseq 1 From Asterisk INVITE (G711U) cseq 2 From NexTone 100 Trying cseq 2 From Asterisk 491 Request Pending cseq 2 From Asterisk ACK cseq 1 From NexTone ACK cseq 2 From NexTone 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone 200 OK (G711U) cseq 1 From Asterisk ACK cseq 1 From NexTone BYE cseq 1 From NexTone 200 OK cseq 1 From Asterisk I see this in the console after the call disconnects: WARNING[24417]: chan_sip.c:1938 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 1 (Critical Response) This fails much more often then it is successful... Anyone have a clue on this??? Stu Sheldon Tech Committee Chairperson S.C.A.L.E. - -- Open up the window Let some air into this room I think I'm almost chokin' From the smell of stale perfume And that cigarette you're smoking 'Bout scared me half to death Open up the window, sucker Let me catch my breath -- Three Dog Night - Mama Told Me Not to Come - Lyrics -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.6 (GNU/Linux) iD8DBQFHPNmWK69Y+xPZrWYRAl+1AJ9131OcSgPiC1GBbZvEDrEZ9NcQ8gCgmGRE 98ZBHiZLGzroI1MP+vJj4wM= =e1fd -END PGP SIGNATURE- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Shared gsm files
On Thursday 15 November 2007 09:15:46 David Ruggles wrote: Does anyone store gsm files on a shared server so multiple asterisk boxes can access the common gsm files? I want to do this so they can be updated easily, but wanted to make sure I wouldn't run in to any unforeseen problems. If anyone has done this could you tell me what you used and if you had any problems? You can share files on any medium (NFS or Samba) read-only just fine. The only real issue with shared files is the coordination of writes and the race condition that results when two hosts try to create a file with the same name. -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TE210P Vs TE220P difference
Dear all anybody have idea of this 2 card and performance vise which one is suggestable ??? PGP Signature-- Satish Patel mobile:- +91-9818875535 http://www.linuxbug.org - Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now.___ --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] Changing audio message to text message
EdPimentl a écrit : Yes, it is call http://www.talktext.com/ -E http://mobiquity.ws http://datr.ws ___ --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 It's not what I wanted to know so I hope you're not sending me this URL to make some pub... I want to replace the audio waiting message (used in queue) by a text waiting message...and not read my voicemail... And I'd like to know if Asterisk is able to send a text message indeed of an audio message...Or at least is it able to send a video message ? -- Anthony Chapellier - MBDSYS SARL 1, centre commercial de la Tour 93120 LA COURNEUVE FRANCE E-mail : [EMAIL PROTECTED] Tel : +33 (0) 143 11 09 14 ou +33 (0) 148 35 20 46 Fax : +33 (0) 148 37 79 28 http://www.mbdsys.com ___ --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] Polycom softkey transfer issue
We upgraded to Asterisk 1.4.13 earlier this week, and now an odd problem has cropped up with our Polycom phones. Basically, performing a transfer via the transfer softkey fails on transfers to extensions beginning with a *. When you try, you just get a fast-busy, and next to nothing on the console (usually a message about the peer being unable to authenticate) This is only affecting our Polycom phones. We have a couple of Aastra's as well on which the issue is not occurring. I can provide packet dumps of a working test with one of the Aastra's and a failed test with one of the Polycom's if their needed, but, right now, I just want to see if anyone has even a wild guess at what might be happening. I'm well and truly stumped on this one. Just to sum up: Polycom phones fail when transferring to an extension prefixed by a *, using the transfer softkey Aastra phones have no such difficulty The only real console output is a mention of the peer being unable to authenticate (which is odd since it's already registered) This did not start occurring until our upgrade to 1.4.13 of Asterisk Thanks for any insight you can provide, Wayne ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files
The softkeys translate fine. Things like redial, new call, call forward, transfer, conference, hold, end call, do not disturb (for DND you have to go through a few more menus) line selection, services. I'll try attaching a screenshot of the softphone I have setup. I've setup the services button so you can browse the local extension directory (based on the sip.conf file) and I also setup a script to generate system speeds dials for all the phones. It also alphabetizes them automatically. I'm hoping to use a nonstandard template to make things like DND a bit more accessible. I just received the 7941 7911g phones from our Cisco rep I'm working on loading the SIP image on those. Oh the other thing I created is a script for auto generation of your SEPmac.cnf.xml file for each phone. You just enter in the mac address the sip extension, password, display name and phone label and the xml file is automatically generated. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olivier Sent: Thursday, November 15, 2007 12:29 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Cisco 7911/7941/7970/7971 Softkey XML Files 2007/11/15, Greg Oliver [EMAIL PROTECTED]: On Thu, 2007-11-15 at 07:31 +0100, Olivier wrote: 2007/11/14, Greg Oliver [EMAIL PROTECTED]: On Tue, 2007-11-13 at 08:44 -0500, Anciso, Roy wrote: Hello List, Does anyone have access to the soft key configuration files for the Cisco 7911/7941/7970/7971 phones? Checked up on the Cisco site and didn't find much up there. Thanks Softkeys running both SCCP and SIP firmware are both sent through the protocols themselves. How ? In SIP mode, is it using RegEvents (rfc3680) ? regards Cisco using RFCs - lol - I wish... Without softkey configuration files, I've heard you cannot translate menus when connecting a Cisco SIP phone to any non-Cisco SIP server. -Greg ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] reload command
I don't think that is the case at all. I believe that all calls will carry on without interruption. Julian Well I learned something new today... thank you. In the past, I could swear that reloading chan_sip on a bridged call would cause me to loose connection. But now, I cannot get even the reload to do this. Good to know. I am running Asterisk SVN-branch-1.4-r89125M and this does not occur. Thanks again Julian. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Problem
Does it help to turn on dtmf log in each servers? On Nov 16, 2007 5:01 PM, 木木 [EMAIL PROTECTED] wrote: I think you haven't capture the packet from the beginning of the call. You must capture the SIP packets. And the wireshark will recognise the packets as RTP. 木木 2007-11-16 发件人： Benjamin Jacob 发送时间： 2007-11-16 12:55:51 收件人： Asterisk Users Mailing List - Non-Commercial Discussion 抄送： 主题： Re: [asterisk-users] DTMF Problem for UDP tcpdump -nnXs 0 udp -i eth0 -w name.cap Btw, a pcap file (created on a linux server using tcpdump) capturing the RTP(udp) traffic opened up in wireshark, wireshark doesn't really format(or recognize) the packets as RTP, unlike the capture done live from a wireshark configured to capture RTP traffic. In the former, wireshark shows up everything as UDP and I have to do a lot of manual parsing to find out the type etc in the packets captured. Am I missing some config on wireshark here? TiA - Ben. ľľ wrote: You can use the tcpdump comand in linux. Like: tcpdump -i eth0 -s 0 -w name.cap And you can open the cap file useing wireshark that is a good 木木 2007-11-16 *发件人：* Doug *发送时间：* 2007-11-16 00:53:15 *收件人：* Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - No *抄送：* *主题：* Re: [asterisk-users] DTMF Problem At 06:42 11/15/2007, =?gb2312?B?xL7Evg==?= wrote: Hi， Could you capture the the UDP package How is this done? in all of your server, Asterisk A, Asterisk B, ser, Asterisk C. And you can find that server who lost the DTMF (RTP EVENT). -- Amy 2007-11-15 -- 发件人： Arun Kumar 发送时间： 2007-11-15 20:30:45 收件人： Asterisk Users Mailing List - Non-Commercial Discussion; SER Users 抄送： 主题： [asterisk-users] DTMF Problem Hi Here is my setup: USER -- PSTN - Asterisk A IAX2 Trunk Asterisk B - SER Asterisk C I'm not able to receive DTMF passed by USER on Asterisk C. All my asterisk boxs are configured with same DTMF type (auto) but no luck. Please help on this issue. Thanks, Arun ___ --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 EMAIL DISCLAIMER : This email and any files transmitted with it are confidential and intended solely for the use of the individual or entity to whom they are addressed. Any unauthorised distribution or copying is strictly prohibited. If you receive this transmission in error, please notify the sender by reply email and then destroy the message. Opinions, conclusions and other information in this message that do not relate to official business of Mascon shall be understood to be neither given nor endorsed by Mascon. Any information contained in this email, when addressed to Mascon clients is subject to the terms and conditions in governing client contract. Whilst Mascon takes steps to prevent the transmission of viruses via e-mail, we can not guarantee that any email or attachment is free from computer viruses and you are strongly advised to undertake your own anti-virus precautions. Mascon grants no warranties regarding performance, use or quality of any e-mail or attachment and undertakes no liability for loss or damage, howsoever caused. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Pass CallerID when call forwards to PSTN?
Hi, Incoming calls to one of my lines are set to ring two internal lines and simultaneously start ringing my cell phone. Something like this: exten = s,1,Dial(SIP/2201SIP/2202IAX2/[EMAIL PROTECTED]),90) The internal lines 2201 and 2202 will both see the callerID for the incoming call, but my cell phone will show the callerID for asterisk, not the calling party. What's the best solution to take the callerID from the inbound call and transfer it to the outbound one? I'm still using v1.2 here. Thanks, Russell. ___ --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] Toshiba DK - Asterisk Integration
Please help me in this regards.. Thanks, Indika. Indika Wasala wrote: Hi All, Interfaces of my PBX are as follows, Toshiba dk28 CO Lines (to telcos) : 12 - (2 free) Digital extensions : 8 - (full) Analog extensions :18 - (full) Toshiba dk280 CO Lines (to telcos) : 8 - (1 free) Digital extensions : 16 - (5 Free) Analog extensions : 16 - (1 free) Toshiba dk8 CO Lines (to telcos) : 4 - (1 free) Digital extensions : 8 - (2 free) Analog extensions : 2 Non of the systems have T1 interfaces and also it seems these systems does not support T1. What you mean is if I need 5 IP phones (sip extensions) I need 5 POTS interfaces. Please advice. Thanks Indika. Tony Plack wrote: Indika, The question of interface depends on how your Strata PBX are connected to the telco currently and what interfaces your Strata supports. If all you have is POTS interfaces to the telco, your integration may be limited because every SIP extension will require a separate POTS line to the Strata. But if you have a T1 interface, you should be able to have trunked lines/multiple extensions. So we need more details. Tony Plack Hi All, I am new to both Asterisk and PBX stuff. I have 3 Tohiba PBXs in 3 separate offices as follows, Toshiba Strata dk28 Toshiba Strata dk280 Toshiba Strata dk8 I need to install 3 Asterisk servers in these 3 locations and integrate them with each of the Toshiba PBX s. This is to give IP Phones/soft phones to the users and to route these VOIP calls through the PBX to POTS. What are the Digium cards I should use in each of these cases and How should I integrate Asterisk with above systems. I read the article in http://www.voipinfo.org/wiki/index.php?page=Asterisk-ToshibaStrata and not sure whether that scenario fits mine. Also it is bit confusing to identify what Digium cards should I need for my cases. Any help is highly appreciated. Thanks, Indika. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] r2 multiframe error - continue
Wll, I think you should have started all this thread by mentioning that. May be libmfcr2 do not support R2 variant in poland. Weeell ;) My mistake.. For you, the quick solution might be just ask your E1 in ISDN-PRI. If you really want to stay with R2 or you have no choice, we can arrange a meeting to start figuring out how to support poland R2 variant or a work-around for it. However I will not have any time before this wednesday. I dont want to stay with R2 but my teleco force this signalling. I will ask them once again. I will also install new elastix and try to change protocolvariant and ...change protocolvariant .. and change protocolvariant ;] Thanks for help once again Have a great weekend. You too - Moy Arkon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Problem
You can use the tcpdump comand in linux. Like: tcpdump -i eth0 -s 0 -w name.cap And you can open the cap file useing wireshark that is a good 木木 2007-11-16 发件人： Doug 发送时间： 2007-11-16 00:53:15 收件人： Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - No 抄送： 主题： Re: [asterisk-users] DTMF Problem At 06:42 11/15/2007, =?gb2312?B?xL7Evg==?= wrote: Hi， Could you capture the the UDP package How is this done? in all of your server, Asterisk A, Asterisk B, ser, Asterisk C. And you can find that server who lost the DTMF (RTP EVENT). -- Amy 2007-11-15 -- 发件人： Arun Kumar 发送时间： 2007-11-15 20:30:45 收件人： Asterisk Users Mailing List - Non-Commercial Discussion; SER Users 抄送： 主题： [asterisk-users] DTMF Problem Hi Here is my setup: USER -- PSTN - Asterisk A IAX2 Trunk Asterisk B - SER Asterisk C I'm not able to receive DTMF passed by USER on Asterisk C. All my asterisk boxs are configured with same DTMF type (auto) but no luck. Please help on this issue. Thanks, Arun ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Building an Asterisk 1.4 RPM.
I'm a little confused. I'd like to build an RPM for Asterisk 1.4. Is it better to modify and use the spec file under redhat/asterisk.spec and run a 'make rpm', OR is it better to build a custom spec file from scratch and use 'rpmbuid -ba' specfile? How do people normally do it? The problem I see with a custom spec file is that since the source is all contained within a tar.gz file, there's no way to interactively run a 'make menuselect' first and customise or remove what you don't need. For example, if I don't do this, the ogg vorbis module is installed by default, and then when I go to install my rpm, there's complaints all round if the ogg vorbis libs aren't already installed. Doug. Get easy, one-click access to your favorites. Make Yahoo! your homepage. http://www.yahoo.com/r/hs ___ --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] r2 multiframe error - continue
Let's start with something basic, try connecting in loop and using protocolvariant=mx,0,4,7 then call yourself. That MUST work. Otherwise you have messed up installing the incorrect libraries, I have seen too many people complaining about the libraries not working and they just forgot to install proper spandsp version or something like that. Other common error is duplicating libraries installed in /usr/local with the ones in /usr/lib Regards On Nov 16, 2007 6:31 AM, Jakub Syrek [EMAIL PROTECTED] wrote: Im using libs from astunicall-1.4.9-0.1.tar.gz at http://www.moythreads.com/astunicall/downloads/ (i have reinstalled asterisk, and libs from this package once again) No one can call me and i cant call out. Man from teleco still have teletransmision error.. No after starting asterisk im getting in full log something like this: [Nov 16 13:22:09] NOTICE[3787] chan_unicall.c: Unicall/17 event Detected [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 - C on [2/DETECTED/Seize ack /Seize ack] [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 R2 prot. err. [2/DETECTED/Seize ack /Seize ack] cause 32772 - Unexpected MF6 signal [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 1001 - [1/IDLE/Idle /Idle ] [Nov 16 13:22:09] NOTICE[3787] chan_unicall.c: Unicall/17 event Protocol failure [Nov 16 13:22:09] ERROR[3787] chan_unicall.c: Unicall/17 protocol error. Cause 32772 [Nov 16 13:22:09] WARNING[3787] chan_unicall.c: MFC/R2 UniCall/17 Channel echo cancel [Nov 16 13:22:09] DEBUG[3787] chan_unicall.c: disabled echo cancellation on channel 17 What can i do?:) Regards Arkon ___ --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 -- Within C++, there is a much smaller and cleaner language struggling to get out. ___ --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] reload command
See inline: Tony Plack wrote: I'm talking about the command: /usr/sbin/asterisk -rx reload Jerry I am not sure why one would need reload these days unless you are making changes to the code in the modules. My guess is that you are doing this after a change to extensions.conf. If this is the case, please try /usr/sbin/asterisk -rx extensions reload or if you need to reload a specific module /usr/sbin/asterisk -rx module reload x Of course if you have a sip call in progress and you call reload on chan_sip, expect that the connection will be broken. I don't think that is the case at all. I believe that all calls will carry on without interruption. Julian ___ --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 __ This email for dotr.com has been scanned by MessageLabs __ ___ --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] Changing audio message to text message
Yes, it is call http://www.talktext.com/ -E http://mobiquity.ws http://datr.ws ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Music on Hold -- Error
Interesting. Is the upgrade difficult? I've not attempt to upgrade our production environment yet. Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nick Brown Sent: Thursday, November 15, 2007 7:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Music on Hold -- Error I posted to the list earlier this week about this very issue. This reinforces my thought that it is a bug in 1.4.7. Since upgrading the box to 1.4.13 the issue resolved itself. I have not opened a issue in the tracker as I hadn't had time to try and replicate the issue. On 16/11/07 5:32 AM, Ryan M. Colbert [EMAIL PROTECTED] wrote: We use Asterisk 1.4.7 on CentOS with Bandwidth.com as our provider and Polycom 330's for endpoints. When one of our end points places a call on hold we get the following in CLI. There is no music on hold provided for the caller. The SIP.CONF entry for our connection to bandwithd.com specifies disallow=all and allow=ulaw. Should there be a similar setting on the user.conf entries? An interesting note is the IP noted in the CLI message below is neither Bandwidth.com nor the end point. Thanks for any help!! CLI Message: [Nov 15 13:21:38] WARNING[11327]: channel.c:2964 set_format: Unable to find a codec translation path from ulaw to unknown [Nov 15 13:21:38] WARNING[11327]: res_musiconhold.c:702 moh_alloc: Unable to set channel 'SIP/4.68.250.148-08d1e7e0' to format 'unknown' Ryan M. Colbert Director of Information Technology Rissman, Barrett, Hurt, Donahue McLain, P.A. 201 E. Pine Street, Suite 1500 Orlando, FL 32801 (407) 517-3105 - Direct Telephone (407) 839-0120 - Main Office (407) 841-9726 - Fax http://www.rissman.com/ http://www.rissman.com/http://www.rissman.com/ ___ --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 Regards, Nick Brown Ipera Communications Pty Ltd Level 1, 9 Denison Street, Newcastle West NSW 2302 PO Box 2115, Dangar NSW 2309 Ü P: +61 2 4910 1000 Ü F: +61 2 4910 1099 Ü ABN: 31 090 964 104 ___ --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] Dumb AGI question
I did some simple AGI programming several months ago. I have a need to use one of those old programs and I'm having a stupid problem. I can't get output to display on the console. I'm sending it to stderr and I've got verbosity set to 10. I know I had it working before so I'm guessing I just forgot some piece of key information. Any suggestions? Thanks, David Ruggles CCNA MCSE (NT) CNA A+ Network EngineerSafe Data, Inc. (910) 285-7200 [EMAIL PROTECTED] ___ --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] dtmf detection
Hi, Below is my case. phoneA (PSTN) phoneB (SIP) phoneC (PSTN) phoneA -- asterisk -- phoneB phoneA (music on hold), phoneB --attended call transfer-- phoneC phoneA --connect-- phoneC after phone B hangup phoneA type some keys in keypad but phoneC always has wrong dtmf detection. In my case, I would like to know any factor that will cause the wrong dtmf detection. ___ --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] Dialing time-out
On Thu, 2007-11-15 at 10:37 -0600, Jim Houser wrote: One of the issues with user devices at the end Asterisk is dialing time out [snip] This clearly separates Asterisk from the traditional TDM platform behavior where a time out can be REAL LONG allowed people to dial at a snail's rate without upsetting the phone system but then immediately out pulsing when a number match is met, regardless if the number match is a 4 digit extension or 7 digit phone number. Actually, this isn't quite correct. With Asterisk, you can define both the response timeout and the digit timeout (the one you specifically mention above) using the TIMEOUT dialplan function. As for having the system immediately dial out once an extension is matched, it's really up to your dialplan. Asterisk will connect to the extension as soon as there's an *unambiguous* match. Point an analog phone at the context below, and I think you'll see what I'm trying to say. (Obviously SIP phones are different than analog, in that they usually send the entire dialed number at once -- if you're using a SIP phone, you may be encountering a dial timeout on your phone, and not in Asterisk.) [dial-timout-test] ; If you dial 1 or 12, Asterisk will wait before connecting, to see ; if you're going to enter the 3 for extension 123 exten = 1,1,SayNumber(1) exten = 12,1,SayNumber(12) exten = 123,1,SayNumber(123) ; If you dial 2, Asterisk will immediate connect you, as there's no ; other possible match in this context. exten = 2,1,SayNumber(2) -- Jared Smith Community Relations Manager Digium, Inc. ___ --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 1.4.14 Released
The Asterisk Development Team has released Asterisk version 1.4.14. This is a regular maintenance release that contains numerous bug fixes across the entire code base. A ChangeLog that lists all changes that were made is available with the release. http://svn.digium.com/view/asterisk/tags/1.4.14/README?view=markup The release is available on downloads.digium.com. It is also available as a patch against the previous release. http://downloads.digium.com/pub/telephony/asterisk/ Thank you for your support! ___ --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] modifying a dialed exension before dialplan processing
I have a phone (a panasonic globalrange phone) which always sends a
fully qualified phone number. That is, for a local Canadian number,
even if I key in 6135551212 it actually sends to asterisk
01116135551212. This means of course, along with normal phones I end
up having twice as many extensions for outdialed numbers.
Is there any way I could canonicalize this down to the more normal
NXXNXX format before I process through all of my dialplan rules?
Effectively it means being able to alter ${EXTEN}.
Is this doable in any way?
b.
___
--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] r2 multiframe error - continue
Hi Jakub, Most countries which used to be part of the iron curtain block, back in the good old days, use the same protocol. Try the Czech variant. It will probably be OK for you. If it works, please report that, and Poland can be added to the list of variants. Steve Jakub Syrek wrote: Im from Poland and there is no pl option, what should i chose? Arkon - Original Message - From: Moises Silva [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, November 16, 2007 7:01 PM Subject: Re: [asterisk-users] r2 multiframe error - continue So, that means it is succeeded for mx protocolvariant. Now, just change the protocolvariant 'mx' to whatever fits your country, change only the country, please leave ANI,DNIS,OPTIONS . If it fails, I think a bug exists in your particular protocolvariant. Let me know the results. On Nov 16, 2007 11:11 AM, Jakub Syrek [EMAIL PROTECTED] wrote: I was testing my system in local loop for protocolvariant mx,3,3(e1 cross cable between two spans). Here are results: testcall Loading protocol mfcr2 Thread for channel 0 MFC/R2 Chan 41: Call control(9) MFC/R2 Chan 41: Unblock MFC/R2 Chan 41: 1001 - [1/BLOCKED /Idle /Idle ] MFC/R2 Chan 41: far_unblocking_expired MFC/R2 Chan 41: local_unblocking_expired Chan 41: -- Far end unblocked! :-) Chan 41: -- Far end unblocked! :-) Chan 41: -- Local end unblocked! :-) Chan 41: -- Local end unblocked! :-) Chan 41: Initiating call MFC/R2 Chan 41: Call control(1) MFC/R2 Chan 41: Make call MFC/R2 Chan 41: Creating a new call with CRN 32769 MFC/R2 Chan 41: 0001 - [1/DIALING /Seize /Idle ] Chan 41: -- Dialing on channel 0 Chan 41: -- Dialing on channel 0 MFC/R2 Chan 41: - 1101 [1/DIALING /Seize /Idle ] MFC/R2 Chan 41: 1 on - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 6 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 1 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 6 off [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: Calling party category 0x0 MFC/R2 Chan 41: 1 on - [2/DIALING /Group III /Category ] MFC/R2 Chan 41: - 5 on [2/DIALING /Group III /Category ] MFC/R2 Chan 41: 1 off - [2/DIALING /Group III /Category ] MFC/R2 Chan 41: - 5 off [2/DIALING /Group III /Category ] MFC/R2 Chan 41: 2 on - [2/DIALING /Group III /DNIS ] MFC/R2 Chan 41: - 1 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 2 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 off [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 3 on - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 on [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: 3 off - [2/DIALING /Group I /DNIS ] MFC/R2 Chan 41: - 1 off [2/DIALING /Group I /DNIS ] Main thread MFC/R2 Chan 41: - 3 on [2/DIALING /Group I /Silent ] MFC/R2 Chan 41: - 3 off [2/DIALING /Group I /Silent ] MFC/R2 Chan 41: 1 on - [2/PROCEED /Group II /Category ] Chan 41: -- Proceeding on channel 0 MFC/R2 Chan 41: - 1 on [2/PROCEED /Group II /Category ] MFC/R2 Chan 41: 1 off - [2/PROCEED /Group II /Category ] MFC/R2 Chan 41: - 1 off [2/PROCEED /Group II /Category ] Chan 41: -- Alerting on channel 0 Chan 41: -- Alerting on channel 0 MFC/R2 Chan 41: - 0101 [1/ALERTING/Await answer /Category ] Chan 41: -- Connected on channel 0 Chan 41: -- Connected on channel 0 Chan 41: -- '*0001*343*123*#' Main thread Main thread Main thread MFC/R2 Chan 41: - 1101 [1/CONNECTD/Answered /Category ] MFC/R2 Chan 41: Far end disconnected(cause=Normal Clearing [16]) - state 0x400 and asterisk log [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/25 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/26 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/27 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/28 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/29 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/30 event Local end unblocked [Nov 16 18:06:51] NOTICE[28848] chan_unicall.c: Unicall/31 event Local end unblocked [Nov 16 18:07:07] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 - 1001 [1/BLOCKED /Idle /Idle ] [Nov 16 18:07:08] WARNING[28848] chan_unicall.c: MFC/R2 UniCall/10 far_unblocking_expired [Nov 16 18:07:08] NOTICE[28848] chan_unicall.c: Unicall/10 event Far end unblocked [Nov
Re: [asterisk-users] modifying a dialed exension before dialplan processing
I have no idea if this would work :
exten = _0111NXXNXX,1,Set(x=${EXTEN:4})
exten = _0111NXXNXX,n,Goto(${x},1)
exten = _0111NXXNXX,n,NoOp( Sorry it didn't work ! )
exten = _0111NXXNXX,n,Hangup()
;
exten = _NXXNXX,1,NoOp( OMG, It worked ! )
exten = _NXXNXX,n,NoOp( continue like other calls )
--
On Nov 16, 2007 7:38 PM, Brian J. Murrell wrote:
I have a phone (a panasonic globalrange phone) which always sends a
fully qualified phone number. That is, for a local Canadian number,
even if I key in 6135551212 it actually sends to asterisk
01116135551212. This means of course, along with normal phones I end
up having twice as many extensions for outdialed numbers.
Is there any way I could canonicalize this down to the more normal
NXXNXX format before I process through all of my dialplan rules?
Effectively it means being able to alter ${EXTEN}.
Is this doable in any way?
b.
--
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] modifying a dialed exension before dialplan processing
exten = _0111NXXNXX,1,Goto(${EXTEN:4},1)
exten = _NXXNXX,1,Dial(
Baji Panchumarti wrote:
I have no idea if this would work :
exten = _0111NXXNXX,1,Set(x=${EXTEN:4})
exten = _0111NXXNXX,n,Goto(${x},1)
exten = _0111NXXNXX,n,NoOp( Sorry it didn't work ! )
exten = _0111NXXNXX,n,Hangup()
;
exten = _NXXNXX,1,NoOp( OMG, It worked ! )
exten = _NXXNXX,n,NoOp( continue like other calls )
--
On Nov 16, 2007 7:38 PM, Brian J. Murrell wrote:
I have a phone (a panasonic globalrange phone) which always sends a
fully qualified phone number. That is, for a local Canadian number,
even if I key in 6135551212 it actually sends to asterisk
01116135551212. This means of course, along with normal phones I end
up having twice as many extensions for outdialed numbers.
Is there any way I could canonicalize this down to the more normal
NXXNXX format before I process through all of my dialplan rules?
Effectively it means being able to alter ${EXTEN}.
Is this doable in any way?
___
--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
