[Asterisk-Users] Asterisk to CCM Message Waiting Indicator
I am trying to setup Asterisk as the voicemail server for Cisco Call Manager. I
have just about everything working except for the message waiting indicator.
I have the following setup in context [ccm] in my extensions.conf file:
;MWI
exten = _2807XXX,1,SetCallerID(${EXTEN:3})
exten = _2807XXX,2,Dial(SIP/[EMAIL PROTECTED])
exten = _2807XXX,3,Answer
exten = _2807XXX,4,Wait,1
exten = _2807XXX,5,Hangup
;MWI
exten = _2817XXX,1,SetCallerID(${EXTEN:3})
exten = _2817XXX,2,Dial(SIP/[EMAIL PROTECTED])
exten = _2817XXX,3,Answer
exten = _2817XXX,4,Wait,1
exten = _2817XXX,5,Hangup
I also have the vm.sh file in /var/lib/asterisk/scripts/vm.sh:
if [ $3 -gt 0 ]; then
CALLFILE=$(cat -EOF1
Channel: Local/281$2
MaxRetries: 1
# Retry in 2 min
RetryTime: 120
WaitTime: 45
Context: ccm
Extension: s
Priority: 1
EOF1)
echo $CALLFILE /var/spool/asterisk/outgoing/$(date
+%Y%mNaVI%M%S)-$1
else
CALLFILE=$(cat -EOF1
Channel: Local/280$2
MaxRetries: 1
# Retry in 2 min
RetryTime: 120
WaitTime: 45
Context: ccm
Extension: s
Priority: 1
EOF1)
echo $CALLFILE /var/spool/asterisk/outgoing/$(date
+%Y%mNaVI%M%S)-$1
fi
Unfortunately, I keep getting these errors on every voicemail that I leave:
Oct 7 14:36:16 NOTICE[17889]: chan_local.c:455 local_alloc: No such
extension/context [EMAIL PROTECTED] creating local channel
Oct 7 14:36:16 NOTICE[17889]: channel.c:2098 __ast_request_and_dial: Unable to
request channel Local/2817001
Oct 7 14:36:16 NOTICE[17889]: pbx_spool.c:243 attempt_thread: Call failed to
go through, reason 0
I know this is probably fairly easy to fix but I'm not exactly sure how the
outbound call files in /var/spool/asterisk/outgoing work. Any ideas as what I
can change. One other question, I have several different voicemail contexts
setup and this is the only one where I need an external notification. Is there
a way to set this up so that I only send these notify calls when there is a
voicemail in this ccm context? Any help would be greatly appreciated.
Thanks,
Brian
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk to CCM
Dinesh, Yes, that was the problem. I had everything setup as None and apparently for inbound calls Calling search space was necessary. Thanks for your help, Brian -Original Message- From: Dinesh Birlasekaran [mailto:[EMAIL PROTECTED] Sent: Tuesday, September 27, 2005 11:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to CCM Hi Brian, Calling search space is correct? Dinesh. On Mon, 26 Sep 2005, Brian J. Rathman wrote: I am currently trying to send calls from Asterisk to Cisco Call Manager 4.0 and vice versa. I have a SIP trunk setup in CCM and I also have an entry in my sip.conf file for CCM. Unfortunately, when I try and send a call to CCM I keep getting: SIP/10.0.0.1-9c18 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 10.0.0.1 I have another SIP trunk setup in CCM pointing to my cisco as5300 gateway. I can send calls to it and they complete, but when I point the route pattern to Asterisk it fails immediatly. Any suggestions? Thanks, Brian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users DISCLAIMER: This email is confidential and may be privileged. If you are not the intended recipient, please delete it and notify us immediately. Please do not copy or use it for any purpose, or disclose its contents to any other person as it may be an offence under the Official Secrets Act. Thank you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk to CCM
No, I have my cisco gateway setup as a SIP trunk in CCM. The calling search area was my problem on inbound call from Asterisk to CCM. The problem on calls from CCM to Asterisk was that I had the inbound context setup incorrectly in sip.conf. -Original Message- From: Greg Oliver [mailto:[EMAIL PROTECTED] Sent: Monday, September 26, 2005 1:54 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to CCM Are you using CCM to operate your gateway with MGCP? If so, I had to change the default timers under CCM advanced setup for Media exchange timers or the call was timing out at 4 seconds. If the setup was complete prior, it worked fine, but after 4 seconds q.931 from CCM would tear down the call.. On Mon, 2005-09-26 at 14:14 -0300, Arnaldo M. Pereira wrote: Have you read http://www.voip-info.org/tiki-index.php?page=Asterisk% 20Cisco%20CallManager%20Integration ? I've followed these steps and I can make calls from a CCM client to Asterisk, but the end point at the Asterisk side can't hear any audio. On Mon, 2005-09-26 at 12:28 -0400, Brian J. Rathman wrote: I am currently trying to send calls from Asterisk to Cisco Call Manager 4.0 and vice versa. I have a SIP trunk setup in CCM and I also have an entry in my sip.conf file for CCM. Unfortunately, when I try and send a call to CCM I keep getting: SIP/10.0.0.1-9c18 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 10.0.0.1 I have another SIP trunk setup in CCM pointing to my cisco as5300 gateway. I can send calls to it and they complete, but when I point the route pattern to Asterisk it fails immediatly. Any suggestions? Thanks, Brian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk to CCM
One way audio in my experience is always a firewall issue. -Original Message- From: Arnaldo M. Pereira [mailto:[EMAIL PROTECTED] Sent: Monday, September 26, 2005 1:15 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk to CCM Have you read http://www.voip-info.org/tiki-index.php?page=Asterisk% 20Cisco%20CallManager%20Integration ? I've followed these steps and I can make calls from a CCM client to Asterisk, but the end point at the Asterisk side can't hear any audio. On Mon, 2005-09-26 at 12:28 -0400, Brian J. Rathman wrote: I am currently trying to send calls from Asterisk to Cisco Call Manager 4.0 and vice versa. I have a SIP trunk setup in CCM and I also have an entry in my sip.conf file for CCM. Unfortunately, when I try and send a call to CCM I keep getting: SIP/10.0.0.1-9c18 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 10.0.0.1 I have another SIP trunk setup in CCM pointing to my cisco as5300 gateway. I can send calls to it and they complete, but when I point the route pattern to Asterisk it fails immediatly. Any suggestions? Thanks, Brian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Arnaldo M. Pereira egghunt at gmail dot com http://ansi-c.org/~arnaldo ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk to CCM
I am currently trying to send calls from Asterisk to Cisco Call Manager 4.0 and vice versa. I have a SIP trunk setup in CCM and I also have an entry in my sip.conf file for CCM. Unfortunately, when I try and send a call to CCM I keep getting: SIP/10.0.0.1-9c18 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Got SIP response 481 Call Leg/Transaction Does Not Exist back from 10.0.0.1 I have another SIP trunk setup in CCM pointing to my cisco as5300 gateway. I can send calls to it and they complete, but when I point the route pattern to Asterisk it fails immediatly. Any suggestions? Thanks, Brian ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Delay after Dial Application is Called
I just recently loaded a new Asterisk server running the CVS code from 12/3/04. I have been running a previous CVS version without any problems from 6/5/04. On the new server I have noticed an issue where when I place a call from one of my snom 190 phones, I see immediatly as I should: -- Executing Dial(SIP/770818-e457, SIP/[EMAIL PROTECTED]|240|T) in new stack then there is a 20 second delay before I see: -- Called [EMAIL PROTECTED] This delay does not exist on my older server. I would assume that there have been significant changes to the Dial application since June, but does anyone know of a parameter I can tweak to fix this? Thanks for your help, Brian ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Transfering Calls
I am having several users complain about not being able to use the # button when dialing into IVR's, etc, because the # key prompts for transfering the call to another extension. Is there a way to still provide transfer capability, but not use the # key? I am using SNOM 200 phones so if anyone has any suggestions, I would greatly appreciate it. Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] 3 Way Calling on Snom Phones and Asterisk
Has anyone been able to get 3way/Conference working with the snom200 and Asterisk. According to the documentation for the phones the option should come up when you have two lines active on the snom phone. Unfortunately, I don't see this option appear and I am now beginning to wonder if this is a limitation of Asterisk. Does anyone have any suggestions? Any help would be greatly appreciated. Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Audiocodes - Asterisk Implementation
Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get the channels to registers with Asterisk, but anytime I try and send a call I receive these error messages: Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 20587: Found Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:11 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:13 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:17 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:17 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:18 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:20 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:24 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:25 DEBUG[1133742896]: chan_sip.c:706 __sip_autodestruct: Auto destroying call '[EMAIL PROTECTED]' I am using CVS version Asterisk CVS-HEAD-06/18/04-11:53:43. I have tried changing just about every config option I can think of in both Asterisk and the Audiocodes box without any success. Any ideas? I have checked the web for documentation on this setup, and all I have found is that some people have it working, but that is about it, no details. Any help would be greatly appreciated. Thanks, Brian ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Audiocodes - Asterisk Implementation
Here are the setting off of the box. It is my understanding that anything over version 4.20 is ok. Any help would be greatly appreciated. Version ID:4.20.299.4192 DSP Type:48104 DSP Software Version:20231 DSP Software Name:105IM4 Flash Version:192 -Original Message- From: Anton Tinchev [mailto:[EMAIL PROTECTED] Sent: Thursday, July 08, 2004 11:57 AM To: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] Audiocodes - Asterisk Implementation Brian J. Rathman wrote: Anyone out there have the AudioCodes MP-108 working with Asterisk? I am able to get the channels to registers with Asterisk, but anytime I try and send a call I receive these error messages: Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:771 __sip_ack: Stopping retransmission on '[EMAIL PROTECTED]' of Response 20587: Found Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:10 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:11 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:13 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:17 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:17 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:18 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:20 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:24 DEBUG[1133742896]: chan_sip.c:6343 handle_request: Ignoring out of order packet 20589 (expecting 20588) Jul 6 15:12:25 DEBUG[1133742896]: chan_sip.c:706 __sip_autodestruct: Auto destroying call '[EMAIL PROTECTED]' I am using CVS version Asterisk CVS-HEAD-06/18/04-11:53:43. I have tried changing just about every config option I can think of in both Asterisk and the Audiocodes box without any success. Any ideas? I have checked the web for documentation on this setup, and all I have found is that some people have it working, but that is about it, no details. Any help would be greatly appreciated. Thanks, Brian Firmwire version? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
