[asterisk-users] X-lite direct sip call - Is it possible?
Hi Guys, Wondering if anyone has tried to make a direct SIP peer to peer call using x-lite without any registrations of any sort. I can't seem to find the setting. Thanks, bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] QUICKLY - What is the command to change .wmv to Asterisk compatible format using ffmpeg? or any other tool?
Hi Guys, I want to test my first video transmission call from Asterisk 1.6 to X-lite softphone. I set videosupport=yes in SIP [general] and I have place a .wmv file in /var/lib/asterisk/sounds/en and did a chown asterisk.asterisk on it. I guess I have to use Playback command for the file and before that I have to convert the file to h.263??!! I just installed ffmpeg (the conversion tool) but does anyone have a quick command to change .wmv file to h.263 or whatever the Asterisk compatible video format is? Thanks a lot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] QUICKLY - What is the command to change .wmv to Asterisk compatible format using ffmpeg? or any other tool?
Thanks but there are tons of unncessary information that come up and nothing specific to asterisk with that type of search. I have already run through those. Anyhow, I won't want to convert anymore but I am wondering if echo() would work and would echo my cam pictuer back to me. I am trying the following: exten = 20,1,playback(beep)exten = 20,n,Record(/tmp/myvideo:wav)exten = 20,n,Hangup exten = 21,1,Answer()exten = 21,n,Background(/tmp/myvideo) * * Problem is that eyeBeam shows my camera on and my picture but on top says, waiting for remote video for ever. So, it seems asterisk doesn't send picture back to me. I have videosupport=yes in sip.conf [general] and I have allow=h263 in sip.conf How can I go about debugging the video transmission? Thanks On Sat, Apr 17, 2010 at 1:07 PM, Steve Totaro stot...@asteriskhelpdesk.comwrote: On Sat, Apr 17, 2010 at 12:31 PM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, I want to test my first video transmission call from Asterisk 1.6 to X-lite softphone. I set videosupport=yes in SIP [general] and I have place a .wmv file in /var/lib/asterisk/sounds/en and did a chown asterisk.asterisk on it. I guess I have to use Playback command for the file and before that I have to convert the file to h.263??!! I just installed ffmpeg (the conversion tool) but does anyone have a quick command to change .wmv file to h.263 or whatever the Asterisk compatible video format is? Thanks a lot http://tinyurl.com/yyr6tvx -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Gurus ONLY - Too complex of an issue
Thanks for the input. Problem was solved by adding transfer=no in zapata.conf For those who need TBCT, then add transfer=yes and facilityenable=yes in zapata.conf. However, if your telco has RLT or TBCT as a value added service and you have not subscribed to it then you will face my problem if transfer is not set to no -Bruce On Mon, Apr 12, 2010 at 11:28 PM, Don Kelly d...@donkelly.biz wrote: The symptoms look like you’re doing TBCT. Unless you’re recording or, for some other reason, want to supervise the call, TBCT is a more efficient use of your PRI as it frees up channels after the transfer. TBCT isn’t available with analog circuits, but is very similar to the analog flash and transfer. I started typing this a while ago and since see that you’re interested in call recording, so you don’t want TBCT. Good news is that you can indicate that you don’t want TBCT in your .conf files. Bad news is that I don’t know how you do it. But you’ve reduced the problem to its simplest form, and someone will respond with exactly what you need to do. And I see you figured out what it takes… --Don -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is restart of span a concern on PRI?
Hi Guys, I have been checking logs and noticed this over the last night. Should I be concerned? and where to look for further details? Sample: [2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/1 successfully restarted on span 1 [2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/2 successfully restarted on span 1 [2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/3 successfully restarted on span 1 [2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/4 successfully restarted on span 1 Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Being attacked by an Amazon EC2 ...
Speaking of all these attacks, are there any good web managed security monitor tools for CentOS out there that can be installed on the system so that it can give us a visual of let's multiple failed attempts against SSH or HTTPd? Something nice that is simple and doesn't eat a lot resources and spits out everything on the screen? Thanks, Bruce On Tue, Apr 13, 2010 at 9:51 AM, Fred Posner f...@teamforrest.com wrote: On Apr 13, 2010, at 8:04 AM, Hans Witvliet wrote: On Tue, 2010-04-13 at 09:47 +0100, Gordon Henderson wrote: On Tue, 13 Apr 2010, Alyed wrote: Think we need some solution WITHIN the Asterisk core. Roderick A. suggested something that looks nice using iptables, some others have pointed out using RBL or fail2ban, but the best would be to have some generic solution not dependant on third party programs. I'd strongly disagree with this. (And I was the OP of this thread and had my home/office network connection taken down due to it) But then, I'm an old worldy Unix sysadmin and the philosophy of having a program do one thing well is still etched into my core... http://en.wikipedia.org/wiki/Unix_philosophy So get asterisk to do what it does well, then get something else that does what you need to do just as well - built-in to Linux are the iptables firewall rules. Use them! They are very effective and do work. (And you have a choice!) I'll agree with you here. Any aditional security within * is fine, but if someone is simply drowning your bandwith, action must be taken at a lower level. Otherwise you endup re-inventing the wheel for D.o.s. attackes for voip, mail, ssh, ldap, http, rsync, (or any other service you might be running) So a proper job for ip(6)tables, imho -- +1 for outside of asterisk. I want something that blocks it before it gets to the Asterisk processes. I've posted a little script on Team Forrest for how I'm blocking the traffic (using a quick perl script, iptables, and cron). The script is at http://bit.ly/cDHlLq ---fred http://qxork.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is restart of span a concern on PRI?
Thanks, I can sleep better now. On Tue, Apr 13, 2010 at 10:02 AM, Doug Lytle supp...@drdos.info wrote: bruce bruce wrote: [2010-04-13 04:31:27] VERBOSE[3844] logger.c: -- B-channel 0/1 successfully restarted on span 1 It's a normal function: *resetinterval*: sets the time in seconds between restart of unused channels, defaults to 3600 minimum 60 seconds. Some PBXs don't like channel restarts. so set the interval to a very long interval e.g. 1 or 'never' to disable *entirely*. http://www.voip-info.org/wiki/view/Asterisk+config+zapata.conf Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Being attacked by an Amazon EC2 ...
Cool. I am just looking over splunk. Isn't that enough by it's own? or is OSSEC needed to give it raw data? I think these two will take quite some time to understand. Anything simpler out there as well? Thanks, Bruce On Tue, Apr 13, 2010 at 10:42 AM, --[ UxBoD ]-- ux...@splatnix.net wrote: - Original Message - Speaking of all these attacks, are there any good web managed security monitor tools for CentOS out there that can be installed on the system so that it can give us a visual of let's multiple failed attempts against SSH or HTTPd? Something nice that is simple and doesn't eat a lot resources and spits out everything on the screen? Thanks, Bruce How about http://www.ossec.net which you could later integrate with http://www.splunk.com/. -- Thanks - Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] PRI Gurus ONLY - Too complex of an issue
Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16. * Dialplan: [zap_bridge] exten = s,1,answer() exten = s,n,Dial(ZAP/g0/416888) * CLI Output: -- Called g0/416888 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap_bridge, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' * * Here is PRI debug: Starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 16 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2967 q931_release: call 32864 on channel 1 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: RELEASE COMPLETE (90) q931.c:3766 q931_receive: call 32864 on channel 1 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Terminator) Message type: RELEASE (77) q931.c:3801 q931_receive: call 32865 on channel 2 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: RELEASE COMPLETE (90) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
Re: [asterisk-users] PRI Gurus ONLY - Too complex of an issue
Please shed some lights if you can see the source of the problem in the debug. The subject was not meant to be a deterrent but rather emphasizing the complexity of issue at hand. As I noted at the bottom of my post, I appreciate any and all input. -Bruce On Mon, Apr 12, 2010 at 4:02 PM, Tim Nelson tnel...@rockbochs.com wrote: - bruce bruce bruceb...@gmail.com wrote: Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. ...etc I was going to respond with some very insightful and helpful information but I'm not a PRI Guru. Sorry, maybe next time. --Tim -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Gurus ONLY - Too complex of an issue
Thanks for the input Don. HmmmI am not understanding the comment here. I am not doing any flash() or transfer() but rather just dial out and native zap bridge should just connect two channels and only hangup both channel when one party hangs up. Here is what should happen: Call comes in and goes to context zap_bridge. [zap_bridge) exten = s,1,answer exten = s,n,Dial(ZAP/g0/1416777) But what happens instead is the moment that 416-777- picks up and PRI debug shows call active state 10 then there is a request to hangup and both channels go down. This is wrong. If one leg of call is SIP, e.g. Dial(SIP/sip_provider/416777) then everything proceeds fine. Also if a channel from an analogue card is use for the second leg, e.g. Dial(ZAP/g1/416777) then native zap bridge still works. I think someone should be able to find something fishy in the PRI debug that I posted. Please help!!! Thanks, Bruce On Mon, Apr 12, 2010 at 4:14 PM, Don Kelly d...@donkelly.biz wrote: It is normal for the PSTN switch to disconnect both channels when a Two B-Channel Transfer is completed successfully. Are the two parties connected? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Monday, April 12, 2010 2:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] PRI Gurus ONLY - Too complex of an issue Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16. * Dialplan: [zap_bridge] exten = s,1,answer() exten = s,n,Dial(ZAP/g0/416888) * CLI Output: -- Called g0/416888 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap_bridge, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' * * Here is PRI debug: Starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 16 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1
Re: [asterisk-users] PRI Gurus ONLY - Too complex of an issue
It just hit me that you are talking about TBCT. I don't think I am doing TBCT as I still want both channels to keep two lines of my PRI occupied. In addition, I would be interested to know how TBCT is done over PRI. I know that this can be done over analogue with flash(). Can you please elaborate a bit so that TBCT is avoided and all calls are bridged at PBX level. Thanks, Bruce On Mon, Apr 12, 2010 at 4:14 PM, Don Kelly d...@donkelly.biz wrote: It is normal for the PSTN switch to disconnect both channels when a Two B-Channel Transfer is completed successfully. Are the two parties connected? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Monday, April 12, 2010 2:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] PRI Gurus ONLY - Too complex of an issue Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16. * Dialplan: [zap_bridge] exten = s,1,answer() exten = s,n,Dial(ZAP/g0/416888) * CLI Output: -- Called g0/416888 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap_bridge, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' * * Here is PRI debug: Starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 16 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2967 q931_release: call 32864 on channel 1 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
Hi Guys, I am sorry if my issue is not related to this but I think it is. I have a PRI with Bell Canada and when I dial in and have the call transfered to a context to dial out and then have those two channels bridged, the call disconnects with cause 16 just exactly as Jay R. Ashworth shows in his CLI output. Bell Canada support RLT or know as 2BCT or TBCT to some but we have not requested that feature. However, we don't care to keep two channels tied up. Is this not possible through PRI? [zap_bridge] exten = s,1,answer exten = s,n,Dial(ZAP/g0/416777) If incoming leg of call is through PRI and outgoing leg is through SIP or analogue ZAP everything works just fine. But the moment Call comes in through PRI and goes out through PRI both channels drop. I should say that call rings the 2nd party and 2nd party sees Caller ID info and when they press Talk then there is the busy signal. I can post all the debug and bore you with it but maybe someone already knows the answer. I have been looking for this for couple of days now and I don't seem to get anywhere with answers. Input is much appreciated. Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Gurus ONLY - Too complex of an issue
Futher check into the PRI debug I am seeing this which actually relates to TBCT and AOC-E error in /usr/src/libpri/pri_facility.c: Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 03] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 0x03 ] Those error codes specifically relate to RLT or TBCT and AOC-E. My question now is, how to avoid Asterisk from doing a TBCT while this is not a TBCT and I want both channels to stay home so I can do call recording. Thanks, Bruce On Mon, Apr 12, 2010 at 8:51 PM, bruce bruce bruceb...@gmail.com wrote: It just hit me that you are talking about TBCT. I don't think I am doing TBCT as I still want both channels to keep two lines of my PRI occupied. In addition, I would be interested to know how TBCT is done over PRI. I know that this can be done over analogue with flash(). Can you please elaborate a bit so that TBCT is avoided and all calls are bridged at PBX level. Thanks, Bruce On Mon, Apr 12, 2010 at 4:14 PM, Don Kelly d...@donkelly.biz wrote: It is normal for the PSTN switch to disconnect both channels when a Two B-Channel Transfer is completed successfully. Are the two parties connected? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Monday, April 12, 2010 2:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] PRI Gurus ONLY - Too complex of an issue Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16. * Dialplan: [zap_bridge] exten = s,1,answer() exten = s,n,Dial(ZAP/g0/416888) * CLI Output: -- Called g0/416888 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap_bridge, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' * * Here is PRI debug: Starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 16 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: DISCONNECT (69
Re: [asterisk-users] PRI Gurus ONLY - Too complex of an issue - SOLVED
Told you it was too complex of an issue :-) I figured to do this in zapata.conf and all is fine now: transfer=no That was the magic two letter which was sending a request for RLT feature on the line. Set transfer to no and all worries gone. Thanks for the input everyone. -Bruce On Mon, Apr 12, 2010 at 10:10 PM, bruce bruce bruceb...@gmail.com wrote: Futher check into the PRI debug I am seeing this which actually relates to TBCT and AOC-E error in /usr/src/libpri/pri_facility.c: Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 03] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 0x03 ] Those error codes specifically relate to RLT or TBCT and AOC-E. My question now is, how to avoid Asterisk from doing a TBCT while this is not a TBCT and I want both channels to stay home so I can do call recording. Thanks, Bruce On Mon, Apr 12, 2010 at 8:51 PM, bruce bruce bruceb...@gmail.com wrote: It just hit me that you are talking about TBCT. I don't think I am doing TBCT as I still want both channels to keep two lines of my PRI occupied. In addition, I would be interested to know how TBCT is done over PRI. I know that this can be done over analogue with flash(). Can you please elaborate a bit so that TBCT is avoided and all calls are bridged at PBX level. Thanks, Bruce On Mon, Apr 12, 2010 at 4:14 PM, Don Kelly d...@donkelly.biz wrote: It is normal for the PSTN switch to disconnect both channels when a Two B-Channel Transfer is completed successfully. Are the two parties connected? --Don Don Kelly PCF Corp People Come First 651 842-1000 888 Don Kell(y) 651 842-1001 fax -- *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *bruce bruce *Sent:* Monday, April 12, 2010 2:22 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] PRI Gurus ONLY - Too complex of an issue Hi Guys, Full PRI installed in Canada with Sangoma A101DE and Asterisk 1.4.21.2, LibPRI 1.4.10. Placing a call into PRI and then transfering that call out to another number. Problem is that the call rings out but the moment the other party pickups both legs of the call are disconnected give Cause code 16. * Dialplan: [zap_bridge] exten = s,1,answer() exten = s,n,Dial(ZAP/g0/416888) * CLI Output: -- Called g0/416888 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap_bridge, s, 2) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' * * Here is PRI debug: Starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect
Re: [asterisk-users] PRI TBCT - Practical Experience, Anybody?
Problem resolved with setting transfer=no in zapata.conf. On Mon, Apr 12, 2010 at 9:14 PM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, I am sorry if my issue is not related to this but I think it is. I have a PRI with Bell Canada and when I dial in and have the call transfered to a context to dial out and then have those two channels bridged, the call disconnects with cause 16 just exactly as Jay R. Ashworth shows in his CLI output. Bell Canada support RLT or know as 2BCT or TBCT to some but we have not requested that feature. However, we don't care to keep two channels tied up. Is this not possible through PRI? [zap_bridge] exten = s,1,answer exten = s,n,Dial(ZAP/g0/416777) If incoming leg of call is through PRI and outgoing leg is through SIP or analogue ZAP everything works just fine. But the moment Call comes in through PRI and goes out through PRI both channels drop. I should say that call rings the 2nd party and 2nd party sees Caller ID info and when they press Talk then there is the busy signal. I can post all the debug and bore you with it but maybe someone already knows the answer. I have been looking for this for couple of days now and I don't seem to get anywhere with answers. Input is much appreciated. Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
There you go. This confirms that SIP signaling determines where the calls should go. I would take their word with a grain of salt specially with their whole support center our of India. No disrespect, but it is bad service overall. -Bruce On Sat, Apr 10, 2010 at 6:32 PM, Joshua Colp jc...@digium.com wrote: - Tarek Sawah tareksa...@hotmail.com wrote: we started with them two days ago .. and we are facing plenty of False Answer cases on several destinations although ppl said they have a policy against FAS.. anyway i don't know i will be looking into another method to send the RTP to another server, The IP address (and port) of where to send audio is negotiated when the call is setup. You can't change it or specify an IP address to use. Even if you did change the IP address you would be sending it to the port associated with the session on the other media gateway. That would just not work. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
out* of india. On Sun, Apr 11, 2010 at 2:26 AM, bruce bruce bruceb...@gmail.com wrote: There you go. This confirms that SIP signaling determines where the calls should go. I would take their word with a grain of salt specially with their whole support center our of India. No disrespect, but it is bad service overall. -Bruce On Sat, Apr 10, 2010 at 6:32 PM, Joshua Colp jc...@digium.com wrote: - Tarek Sawah tareksa...@hotmail.com wrote: we started with them two days ago .. and we are facing plenty of False Answer cases on several destinations although ppl said they have a policy against FAS.. anyway i don't know i will be looking into another method to send the RTP to another server, The IP address (and port) of where to send audio is negotiated when the call is setup. You can't change it or specify an IP address to use. Even if you did change the IP address you would be sending it to the port associated with the session on the other media gateway. That would just not work. -- Joshua Colp Digium, Inc. | Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI - Native ZAP bridge fails - Is this my patch?
Hi Guys, Has anyone experienced this? Can I have a PRI guru weigh in on this? Thanks, Bruce On Sat, Apr 10, 2010 at 3:46 PM, bruce bruce bruceb...@gmail.com wrote: Hi Guys, I am calling out 416-999- on Channel 1 of PRI and then calling 416-999- on Channel 2 of PRI. When the two channels are going to be ZAP native bridged, both channels hangup and CLI show PRI cause (16). Asterisk Verbose *(Channel 1 already connected to party)*: -- Requested transfer capability: 0x00 - SPEECH -- Called g0/416999 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap-bridge, s, 8) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Here is PRI debug, starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 16 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2967 q931_release: call 32864 on channel 1 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: RELEASE COMPLETE (90) q931.c:3766 q931_receive: call 32864 on channel 1 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Terminator) Message type: RELEASE (77) q931.c:3801 q931_receive: call 32865 on channel 2 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: RELEASE COMPLETE (90) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null System Info: *Bell Canada PRI* *Asterisk 1.4.21.2 * *Lib PRI 1.4.10* Is this my patch? https://issues.asterisk.org/view.php?id=7494 Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http
[asterisk-users] PRI - Native ZAP bridge fails - Is this my patch?
Hi Guys, I am calling out 416-999- on Channel 1 of PRI and then calling 416-999- on Channel 2 of PRI. When the two channels are going to be ZAP native bridged, both channels hangup and CLI show PRI cause (16). Asterisk Verbose *(Channel 1 already connected to party)*: -- Requested transfer capability: 0x00 - SPEECH -- Called g0/416999 -- Zap/2-1 is proceeding passing it to Zap/1-1 -- Zap/2-1 is ringing -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 -- Channel 0/1, span 1 got hangup request, cause 16 -- Hungup 'Zap/2-1' == Spawn extension (zap-bridge, s, 8) exited non-zero on 'Zap/1-1' -- Hungup 'Zap/1-1' Here is PRI debug, starting just before Channel two is connected until both channels are disconnected *(maybe FACILITY 98 is of interest?!)*: Message type: CONNECT (7) q931.c:3626 q931_receive: call 32865 on channel 2 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: CONNECT ACKNOWLEDGE (15) -- Zap/2-1 answered Zap/1-1 -- Native bridging Zap/1-1 and Zap/2-1 Protocol Discriminator: Q.931 (8) len=27 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: FACILITY (98) [1c 14 91 a1 11 02 01 06 06 07 2a 86 48 ce 15 00 08 30 03 02 01 61] Facility (len=22, codeset=0) [ 0x91, 0xA1, 0x11, 0x02, 0x01, 0x06, 0x06, 0x07, '*', 0x86, 'H', 0xCE, 0x15, 0x00, 0x08, '0', 0x03, 0x02, 0x01, 'a' ] PROTOCOL 11 A1 0011 (CONTEXT SPECIFIC [1]) 02 0001 06 (INTEGER: 6) 06 0007 2A 86 48 CE 15 00 08 (OBJECTIDENTIFIER: 2a 86 48 ce 15 00 08) 30 0003 (SEQUENCE) 02 0001 61 (INTEGER: 97) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: DISCONNECT (69) [08 02 80 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: User (0) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Processing IE 8 (cs0, Cause) q931.c:3826 q931_receive: call 32864 on channel 1 enters state 12 (Disconnect Indication) -- Channel 0/1, span 1 got hangup request, cause 16 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Active, peerstate Connect Request q931.c:3015 q931_disconnect: call 32865 on channel 2 enters state 11 (Disconnect Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: DISCONNECT (69) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request q931.c:2967 q931_release: call 32864 on channel 1 enters state 19 (Release Request) Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 96/0x60) (Originator) Message type: RELEASE (77) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] -- Hungup 'Zap/1-1' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 96/0x60) (Terminator) Message type: RELEASE COMPLETE (90) q931.c:3766 q931_receive: call 32864 on channel 1 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 97/0x61) (Terminator) Message type: RELEASE (77) q931.c:3801 q931_receive: call 32865 on channel 2 enters state 0 (Null) NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 97/0x61) (Originator) Message type: RELEASE COMPLETE (90) [08 02 81 90] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Normal Clearing (16), class = Normal Event (1) ] NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null System Info: *Bell Canada PRI* *Asterisk 1.4.21.2 * *Lib PRI 1.4.10* Is this my patch? https://issues.asterisk.org/view.php?id=7494 Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
Just a week ago, I have been in the same situation. Provider was changing from Cisco gateways to I think Nextone and hence provided me many IPs. I found out that the media IPs don't matter and just played around with my NAT settings and all calls can go through just fine by using simply: host=111.111.111.111 and the 111.111.111.111 is just their SIP signaling IP. Their gateway will then transfer asterisk to proper gateways for media. Just give it a try; it should work. But my efforts on finding anything regarding this failed on Google as well. P.S. the voip provider name starts with a T and end with A. Regards, Bruce On Sat, Apr 10, 2010 at 11:50 AM, Tarek Sawah tareksa...@hotmail.comwrote: Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk script to repeat dial of a number
LOLthis is just what I was facing 4 days ago. Unfortunately, Asterisk doesn't provide a software feature in Zaptel to do a BUSY. But people on the list suggest that one should call the telephone company and ask them to busy it. If you have the resource and don't mind the bill of calling the bad line with another line (which is still not full proof because someone else could be calling during that time) then check into spool files and do a little bashscript to run in put files in /var/spool/asterisk/outgoing for calls every two minutes. Oh, if you have access to the box, short-circuit the telco line at the telco demarc. http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out-Bruce 2010/4/10 Shaun Wingrin voi...@gmail.com Say, I'm looking for a simple way to dial a number repeatedly for two minutes at a time. The purpose is to busy up a faulty analogue line in an incoming hunt group. Tx Shaun -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling
Oh, I see. I haven't done a lot of testing on this new IP since the change of gateways happened but I did Canada calls and they go fine. However, this exact provider lies down to their teeth when it comes to problems of call quality and calls not routing. They never accept faults. They even have problems sending calls to Canada and USA. They failed to pass calls to India as well over times. I had a funny issue where they were blocking one specific area code in USA without even telling us. It was just a regular area code. They told me it was blocked but I know it was a lie because they wanted to cover their a$$ as the route was down and it wasn't blocked. I doubt the problem is with sending calls to different media gateway as I think SIP signals take care of that. Just like canreinvite feature. But I reserve the right to be wrong. -Bruce On Sat, Apr 10, 2010 at 4:45 PM, Tarek Sawah tareksa...@hotmail.com wrote: you got the name EXACTLY! i already am doing what you suggest but facing problems with some destinations and they claim that the problem is with my Asterisk server not their routes! -- AHD Tarek Sawah Integrated Digital Systems CCNA, MCSE, RHCE, VoIP Syria: +963 944 618286 USA: +1 347 562 2308 Date: Sat, 10 Apr 2010 15:50:52 -0400 From: bruceb...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Sending RTP media to a different server than SIP Signaling Just a week ago, I have been in the same situation. Provider was changing from Cisco gateways to I think Nextone and hence provided me many IPs. I found out that the media IPs don't matter and just played around with my NAT settings and all calls can go through just fine by using simply: host=111.111.111.111 and the 111.111.111.111 is just their SIP signaling IP. Their gateway will then transfer asterisk to proper gateways for media. Just give it a try; it should work. But my efforts on finding anything regarding this failed on Google as well. P.S. the voip provider name starts with a T and end with A. Regards, Bruce On Sat, Apr 10, 2010 at 11:50 AM, Tarek Sawah wrote: Greetings list i'm trying to connect with a VoIP provider for termination.. and they have offered us three servers to connect with one SIP Signaling server and Two Media servers .. googled for a week and didn't find a way to do this.. so my question. is it possible to be done? Asterisk server 1.4.26.3 _ The New Busy is not the too busy. Combine all your e-mail accounts with Hotmail. http://www.windowslive.com/campaign/thenewbusy?tile=multiaccountocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_4 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ The New Busy is not the old busy. Search, chat and e-mail from your inbox. http://www.windowslive.com/campaign/thenewbusy?ocid=PID28326::T:WLMTAGL:ON:WL:en-US:WM_HMP:042010_3 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
I really like the idea. I will try to ask. I don't know if they will be able to do that easily though. They ask a week or two for any changes to the hunt programming. Thanks, Bruce On Thu, Apr 8, 2010 at 3:29 PM, Edo edo.eku...@gmail.com wrote: Hello.. maybe you can just have the telco do an immediate forward of that number to the fifth number in the hunt group until it is fixed... On Thu, Apr 8, 2010 at 1:15 PM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 8 Apr 2010, John Novack wrote: A simple short on the pair will fix that, though that would require you to be on site, not always an option Would sacrificing a spare line cord (cut, strip, twist together) be an option for the on-site staff? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ekunwe EDO Network Services Tel: 601.497.3932 Fax: 601.500.6990 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
Thanks guys for all the input. I have just noticed that the solution doesn't work for me because the 20 lines are in a hunt. And the line in problem is actually the 4th line and not the 1st. So, for incoming calls, if I have more than 3 calls the 4th one will keep ringing for ever and it won't go to the 5th line unless there is a busy on it. Is there anyway I can put a busy voltage on this line without ramping up a big bill? If the line status shows busy then both of incoming and outgoing calls will use the next line available. I am trying to go this route for couple of days until I get the telco to drop by or if the darn rain stops. Thanks, Bruce On Wed, Apr 7, 2010 at 2:56 PM, Danny Nicholas da...@debsinc.com wrote: Since this is hopefully a temporary problem, it would be simpler to simply do zap destroy channel 1 from a CLI prompt. But yes, the ; in the conf will comment these lines until you undo it. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle Sent: Wednesday, April 07, 2010 1:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls bruce bruce wrote: Can I simply put ; in zapata.conf like this to seclude the first zap line from getting calls in or out? I'm not familiar with FreePBX, but I'd say that's logical. Make the change and then from a console type zap show channels, only 2 though 20 should be showing. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
I am not sure if unplugging line from card would work as it's still in a hunt and calls will keep coming through that number and won't fall over to next line unless there is a BUSY on the line. There is no timeout; it's a hunt on BUSY. Plus, I don't have site access for two days :-) For calls out I give them a funny workaround of using another set to call out and not get audio and then use another phone to call so that a different channel is used. They are happy. Since, I been nagging to them to move to PRI because rain keeps brining their lines down all the time. I can't check zaptel disable of the line now as it nears 9:00 A.M. operation time. I will try that later in the day. I am amazed there is not much control to the lines in situations like this. Thanks for the inputs. On Thu, Apr 8, 2010 at 8:43 AM, Doug Lytle supp...@drdos.info wrote: bruce bruce wrote: Is there anyway I can put a busy voltage on this line without ramping up a big bill? If the line status shows busy then both of incoming and outgoing calls will use the next line available. Isn't it as simple as unplugging that phone line from the card? Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
Thanks for the input. Yep, a busy feature on zaptel is an absolute necessary. See, this is a sort of problem that comes back to everyone and goes away quickly, hence the feature wasn't developed probably. But it will make a great addition and will help people in situations like this. On Thu, Apr 8, 2010 at 9:04 AM, Jeff LaCoursiere j...@jeff.net wrote: On Thu, 8 Apr 2010, bruce bruce wrote: I am not sure if unplugging line from card would work as it's still in a hunt and calls will keep coming through that number and won't fall over to next line unless there is a BUSY on the line. There is no timeout; it's a hunt on BUSY. Plus, I don't have site access for two days :-) Nope - unplugging a line that is in a hunt will result in Ring-No-Answer. Ditto for previous advice to destroy the zap channel or to leave it out of the zaptel configuration. You need to busy out that line. You can only do this onsite as far as I know. Or maybe run a script that continually takes that channel offhook and dials something benign... For calls out I give them a funny workaround of using another set to call out and not get audio and then use another phone to call so that a different channel is used. They are happy. Since, I been nagging to them to move to PRI because rain keeps brining their lines down all the time. I can't check zaptel disable of the line now as it nears 9:00 A.M. operation time. I will try that later in the day. I am amazed there is not much control to the lines in situations like this. I totally agree. A busy out application would be a wonderful addition :) I complained about this a few years back... in the meantime, when I need to do such a thing, I busy it out by shorting it at the block. j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
Indeed the telco has no interest in changing the cable, and by the time they send someone to look at the cable it's a sunny day and everything dried out. Hence the order for PRI. Can't wait to fire it up tomorrow. But, taking this number out of hunt is not so much of an option now as it will cost and take two weeks. LAZY anyhow, I will see if I can get to busy the line. Though, I wish there was something with zaptel driver that could do the busy. As it may not be an option for everyone. Thanks for all the input, Bruce On Thu, Apr 8, 2010 at 1:04 PM, John Novack novacks...@gmail.com wrote: Doug Lytle wrote: Jeff LaCoursiere wrote: On Thu, 8 Apr 2010, bruce bruce wrote: Nope - unplugging a line that is in a hunt will result in Ring-No-Answer. Ditto for previous advice to destroy the zap channel or to leave it out of Our telecom guy said, that when you call the line in for repair, that you need to request them to busy out that line for the duration. Doug YES- You need the local Telco to either busy out the line, or remove from the hunt group. This solves the incoming problem You also need to do the same in your Zapata for outgoing calls, simply put that channel in an unused group. I do hope you hunt from last to first for outgoing, and the telco hunts from first to last for incoming. This will reduce, but not eliminate head on collisions ( glare ) Since this seems related to rain/moisture, trouble is probably external, but if your telco's are anything like the big boys in the US, there will be little interest in repairing the cable trouble, wherever it might be OSP repair is of little interest to the big boys any more John Novack John Novack Checked by AVG - www.avg.com Version: 9.0.801 / Virus Database: 271.1.1/2797 - Release Date: 04/07/10 14:32:00 -- Dog is my co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
Not really when you got call center people who deal with makeup goods :-) and their manager can only break things. I can't trust them anywhere near the server. Let alone me telling them which cable to short on the bix. I would presist for Digium to come up with something that would allow soft short circuit :-) hopefully they hear me. On Thu, Apr 8, 2010 at 2:15 PM, Steve Edwards asterisk@sedwards.comwrote: On Thu, 8 Apr 2010, John Novack wrote: A simple short on the pair will fix that, though that would require you to be on site, not always an option Would sacrificing a spare line cord (cut, strip, twist together) be an option for the on-site staff? -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] D-Channel Span Up without Down
HahahahaI definitly agree with Steve. On Wed, Apr 7, 2010 at 11:44 AM, Steve Totaro stot...@first-notification.com wrote: On Wed, Apr 7, 2010 at 11:26 AM, Jason Walker jason.wal...@amgsrv.comwrote: I am getting a bunch of Primary D-Channel on span 1 up but there was not a down message before that. Is this normal? Confidentiality Statement Notice: This email is covered by the Electronic Communications Privacy Act, 18 U.S.C. 2510-2521 and intended only for the use of the individual or entity to whom it is addressed. Any review, retransmission, dissemination to unauthorized persons or other use of the original message and any attachments is strictly prohibited. If you received this electronic transmission in error, please reply to the above-referenced sender about the error and permanently delete this message. Thank you for your cooperation. No, the font size is not normal. Thanks, Steve T -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] URGENT - How to exclude one ZAP channel for outgoin and incoming calls
Hi Guys, Currently, I have a Sangoma A400 installed with 20 ZAP PSTN analogue lines. The first line is giving me problems due to rain (probably coroded line). My server using FreePBX dials out with g0 (group 0 which includes all 20 lines) and it happens that the bad line is the very first line. Can I simply put ; in zapata.conf like this to seclude the first zap line from getting calls in or out? ;context=from-zaptel ;group=0 ;signalling = fxs_ks ;channel = 1 context=from-zaptel group=0 signalling = fxs_ks channel = 2 Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.6.1/1.6.2 binary packages
Thanks for the update Jason, How do the upgrades work if v1.6.0 is already install and one wants to upgrade to 1.6.2 (once it's available)? yum upgrade asterisk* ??? Thanks On Mon, Apr 5, 2010 at 11:37 AM, Jason Parker jpar...@digium.com wrote: Pablo Ruiz wrote: Hello, Does anyone know when we will see asterisk 1.6.1 (and/or 1.6.2) binary packages at packages.asterisk.org http://packages.asterisk.org? Greets. Packages for 1.6.2 will be available Real Soon Now. It's near the top of my short list. They exist, and are sitting in a(n internal) testing repository. Mostly, I just need to make sure upgrades go smoothly. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] call files in 1.6
Yes, so this works (maybe safer than read=all and write=all): read = system,call,command,agent,user,*originate* write = system,call,command,agent,user,*originate* I wasted probably a week on this - thanks to no documentation back in the days with v1.6. -Bruce On Mon, Apr 5, 2010 at 1:50 PM, Tilghman Lesher tles...@digium.com wrote: On Monday 05 April 2010 11:31:04 Jonathan Addleman wrote: Jerry Geis wrote: I just switched from 1.4.30 to 1.6.2 I initiated a call file - same way in 1.4.30 and nothing happened. I was not aware of changes in the call file to 1.6.2? I was watching the cli and no error showed or anything. In the manager.conf I have things setup. [MyDial] secret= permit=127.0.0.1/255.255.255.0 read = system,call,command,agent,user write = system,call,command,agent,user I noticed the same thing - i think something about the permissions has changed, because when I set it to read=all, write=all, it started working again. Haven't dug around enough to find out exactly what's up though. The originate command requires the originate permission. This is detailed in the UPGRADE.txt file (you _did_ read that file thoroughly, didn't you?). -- Tilghman -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Access denied for user 'a2billinguser
I would suggest you try this. It works: http://a2billing2asterisk.googlepages.com On Mon, Apr 5, 2010 at 5:51 PM, Daniel Abreu dlab...@gmail.com wrote: Hi guys. I am facing this problem here, using a2billing. error: 'Access denied for user 'a2billinguser'@'localhost' (using password: YES)' I am following this step by step http://www.asterisk2billing.org/cgi-bin/trac.cgi/wiki/Installation%20Guideand wend i get into the point that i have to Create a2billing database i am getting this message above. I even try to remove the data base and start fresh aging , and still the error , try to change the a2billinguser password on mysql for 1234 using phpmyadmin and still same error. I really don't know how to proceed, does some one have any idea? Thanks -- Daniel Abreu -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.6.1/1.6.2 binary packages
That's why I said Digium can do a better job with keeping up with the updates. I think 1.6+ versions are non-existent. On Sun, Apr 4, 2010 at 9:54 AM, Pablo Ruiz pablo.r...@gmail.com wrote: Maybe i'm wrong, but http://packages.asterisk.org/centos/5/current/ only has asterisk 1.6.0.* packages.. Where are those 1.6.1/2 rpm's you are talking about?? On Sat, Apr 3, 2010 at 2:28 PM, bruce bruce bruceb...@gmail.com wrote: RPMs for CentOS already exist. Though, I agree with better notification/documentation for these and the keeping up with the updates. On Sat, Apr 3, 2010 at 8:14 AM, Pablo Ruiz pablo.r...@gmail.com wrote: Hello, Does anyone know when we will see asterisk 1.6.1 (and/or 1.6.2) binary packages at packages.asterisk.org? Greets. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] asterisk 1.6.1/1.6.2 binary packages
RPMs for CentOS already exist. Though, I agree with better notification/documentation for these and the keeping up with the updates. On Sat, Apr 3, 2010 at 8:14 AM, Pablo Ruiz pablo.r...@gmail.com wrote: Hello, Does anyone know when we will see asterisk 1.6.1 (and/or 1.6.2) binary packages at packages.asterisk.org? Greets. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] a2billing wont pass the number
I think you have caller ID update set to Yes and A2Billing first asks you to: Enter your Caller ID number and then it asks you: Enter your destination number while you mistake both for destination number. Otherwise, I am confused by the title of your question that your caller id doesn't pass and that the message content is not related to it. -Bruce 2010/3/30 Juan E. Rodríguez jerdg...@gmail.com When you say 'a2billing' won't pass the number, you mean you are calling to an IVR or something like that. And when did you dial you destination number twice??? Saludos, Juan E. Rodríguez -Original Message- From: Nathanial Allan nathanial.al...@gmail.com Date: Tue, 30 Mar 2010 13:08:24 To: asterisk-users@lists.digium.com Subject: [asterisk-users] a2billing wont pass the number I am running into an issue with A2Billing. I will explain first of all that everything else works! the system is 90% complete its just this one small problem I am running into. So my problem is that when I place a call, 1. I dial my number that I want and A2Billing gets activated 2. it asks for my pin, upon successful entry of my pin A2Billing then 3. prompts me for my phone number then 4. The call goes out (and actually connects for the record) So I am entering my destination phone number twice which is not the worst thing that can happen, though it is a little annoying Any light that you can shine on this problem would be greatly appreciated as I have been working on it for too long now and I want to get a product! Thank You NallaN -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk system for church call center
SugarCRM and the church. This sounds just like a business; one that doesn't like to call itself a business but employees tactics. I suggest providing them with a solid cisco system with 100s of thousands dollars in cost where they will have less money left to do bad things to world. Asterisk is too good for a church :) On Wed, Mar 31, 2010 at 3:32 PM, Lenz Emilitri lenz.lo...@gmail.com wrote: We have a lot of clients who run small call centers based on Trixbox, and seem to be pretty happy with them. Have a look here: http://queuemetrics.com/manuals/QM_Trixbox-chunked/ Thanks l. 2010/3/31 Frank Church voi...@googlemail.com On 29 March 2010 21:46, Frank Church voi...@googlemail.com wrote: I have been asked by my church to recommend a VoIP system which can do the following. They do internet radio shows which are sometimes broadcast on radio. They are looking for a system which does the following for about 5 agents, exactly as they have described it. 1. Take incoming calls 2. Put them on hold if there is no one to handle the call immediately, or transfer them to an available agent 3. Take down their details, and number, (if this can be retrieved and saved from the caller id, thats better) 4. Get them to hold on after taking their details if they still want to hold 5. Call them back when the backlog is cleared up. I have a fairly good grasp of the hardware and programming part of Asterisk, having compiled it more than a few times and implemented A2Billing phone card and call shop system with it. But the type of software suited to the Call Center side is where my knowledge gap lies. I am looking for solutions based on the usual Asterisk distributions like AsteriskNow, trixbox, elastix etc, whether ready packaged or requiring additional customization. The matter of whether they will use soft phones, or regular phones with headsets is also something to consider. Soft phones with good GUI's may be preferred if more cost effective for them, although my personal preferences are with hard phones. Any recommendations - the ease of software for the end users is the main thing for me, and integration with the database for taking customers details is the main thing for me. One of the distributions with SugarCRM comes to mind here. Sorry for cross-posting, but ready made and commercially supported systems are not ruled out, if they come within their budget. Regards Frank Church After there response I will go with some of ready made Asterisk distributions, then consider to go for a commercial supported versions if they do not meet the churches needs. Thanks Frank -- Loway - home of QueueMetrics - http://queuemetrics.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Classic NO AUDIO problem - DD-WRT and NAT forwarding - HELP PLEASE!
Hi Everyone, I have tried to set the box to DMZ and also tried to port forward 5060 TCP/UDP and 1/2 UDP to the server IP but it's no use and there is a no audio issue. I am pretty certain it's a NAT issue as the sip call establishes. I also made a succesful IAX2 call through IAX trunking and zap lines on this server but sip doesn't work. I register to an extension but even dialing *97 for voicemail wont' give me any audio. Picture posted here shows my DD-WRT NAT setting: *http://tinypic.com/r/21cuqlu/5* Any input will be much appreciated. This is running latest PBXinaFLASH (which has FreePBX) and I tried using externip=x.x.x.x/255.255.248.0 in /etc/asterisk/sip_nat.conf but it was of no use. Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk general Timeout for digits
Hi Guys, I have a need to alter the general timeout in Asterisk. I am wondering if this is something that is hard coded into Asterisk code or if there is a parameter that can be set somewhere. For outbound, I am using x. and hence unless I append a # sign, I would have to wait maybe 5 seconds or so for the call to go through. Is there anywhere in Asterisk that I can change this 5 seconds to let's say 1 second? I understand that there might be the risk of dialing the number unfinished but that's okay with me. Also, for my situation, I can't use specific dial-plans so please guide me to the general timeout parameter if it exists. Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP signal through one IP and media through different IPs
Hi Everyone, I have a provider who is asking me to send SIP signals through 111.111.111.111 and then media through Media 1: 222.222.22.222 and Media 2: 244.244.244.244. This provider authenticates by IP and I think is using Sonus gear and hence they have some load balancer or something... I have always simply done this to work it out: host=111.111.111.111 peer=type and everything worked. But now when I do that I have no audio with call established. I think it's a problem of me not assigning the media IPs. How can I add those to the trunk settings? Thanks, Bruce -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk general Timeout for digits
Thanks for the input. I am using A2Billing and it takes long time to authenticate PIN number and to dial destination number. If # sign is used then it's a different story and it goes through quick. -Bruce On Sat, Mar 20, 2010 at 11:16 AM, Zeeshan Zakaria zisha...@gmail.comwrote: As soon as the dialed number matches one of the dial patterns defined in extensions.conf or its included files, asterisk starts dialing it. The wait you have is probably from the trunk provider's side because by default asterisk doesn't start playing the ring tone unless it gets acknoledgement from the provider's side indicating that the call is successfully going through. But even before the above process starts, sip soft phones have their own dialing patterns and timeout values. As soon as your dialed number matches one of them, it is sent to asterisk which does the above. So first you'll have to check your sip phone's dialout pattern and timeout values. -- Zeeshan A Zakaria On 2010-03-20 10:58 AM, Doug Lytle supp...@drdos.info wrote: bruce bruce wrote: For outbound, I am using x. and hence unless I append a # sign, I would ha... You really do need to give us a snippet of the outbound code. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
