[asterisk-users] g729 codec over SIP Trunk between CCM and Asterisk
Hello, My problem is, outgoing calls (from asterisk to CCM) work fine but incoming (from CCM to Asterisk) does not work because of CCM is trying to use g729 over SIP trunk. I have found that link after a quick search. Problem is the same as in link below (However my Asterisk version is 1.8.13) and solution seems to have H323 trunk between CCM and Asterisk for using g729 codec. The post was written in 2006. Is there any better solution since that time ? Thanks for reading. link : g279 codec over SIP Trunk between CCM and Asteriskhttps://supportforums.cisco.com/message/1072037 -- _ -- 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] Question about directmedia or canreinvite in sip.conf
Hello, I have a question about directmedia or canreinvite, I have experience that whatever I set directmedia=yes or no. After I run sip show settings. all settings looks the same. My question is how I could make sure from sip show settings that my directmedia configuration is applied. Thanks -- _ -- 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] 回覆︰ Question about directmedia or canreinvite in sip.conf
Hi, If you use rfc2833 and set directmedia=yes, diect media will not work. You must set other value like SIP info in order to make diectmedia work ... Regard/chui kingh man 寄件人︰ Shitian Long longst...@gmail.com 收件人︰ asterisk-users@lists.digium.com 傳送日期︰ 2013年01月17日 (週四) 7:27 PM 主題︰ [asterisk-users] Question about directmedia or canreinvite in sip.conf Hello, I have a question about directmedia or canreinvite, I have experience that whatever I set directmedia=yes or no. After I run sip show settings. all settings looks the same. My question is how I could make sure from sip show settings that my directmedia configuration is applied. Thanks -- _ -- 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] How to exclude non-queue calls from recording ?
Hi, Let say we have a call center from which agents get calls from both on-queues and off-queues calls (ie calls passing through queues or direct calls non passing through queues). Regulation here specify prior consent before recording call. How can I best enforce this compliance ? What would you suggest ? I have thought about the following solution: Have a cron script that, reading CDR files, checks if a call both did not enter a queue and has a recording file, if positive, remove this recording file. Cheers -- _ -- 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] How to give users the capability to set CDR userfield for some calls
Hello,
To my surprise, with asterisk 1.8 (I've not tried with other versions), it
seems you cannot set CDR's userfield from within a dialplan macro called by
dynamic features.
See :
testfeature = *321,self/callee,Macro,toto
[macro-toto]
exten = s,1,Verbose(0,Into macro-toto with CDR(src) set to ${CDR(src)})
exten = s,n,Set(CDR(userfield)=foobar)
I'm planning to use this feature to let users mark in CDR an ongoing call
as malicious or important or whatever.
Any hint ?
Regards
--
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Re: [asterisk-users] How to give users the capability to set CDR userfield for some calls
Possibly switch to using subroutines instead of Macros. Macros are being
deprecated in place of subroutines.
Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208
From: Olivier oza_4...@yahoo.fr
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com,
Date: 01/17/2013 10:29 AM
Subject:[asterisk-users] How to give users the capability to set
CDR userfield for some calls
Sent by:asterisk-users-boun...@lists.digium.com
Hello,
To my surprise, with asterisk 1.8 (I've not tried with other versions), it
seems you cannot set CDR's userfield from within a dialplan macro called
by dynamic features.
See :
testfeature = *321,self/callee,Macro,toto
[macro-toto]
exten = s,1,Verbose(0,Into macro-toto with CDR(src) set to ${CDR(src)})
exten = s,n,Set(CDR(userfield)=foobar)
I'm planning to use this feature to let users mark in CDR an ongoing call
as malicious or important or whatever.
Any hint ?
Regards--
_
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http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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--
_
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Re: [asterisk-users] Getting UDPTL (SIP): Transmission error: Resource temporarily unavailable
I'm not using the DHCP server configuration and IP addresses assigned in the network are manual and there are no clashes found in the network. The version of Asterisk I'm using is 10.4.2. I think there might be some issues in this version perhaps I may try to upgrade to 10.12. UDPTL (SIP/10.3.22.6-0ad6): Transmission error to 10.3.22.6:18428: Resource temporarily unavailable. Due to above message, it is badly effecting the V-GW and later I need to restart the Asterisk service. Any thoughts on this? Date: Thu, 17 Jan 2013 15:30:18 +1300 From: Pete Mundy p...@fiberphone.co.nz Subject: Re: [asterisk-users] Getting UDPTL (SIP): Transmission error: On 17/01/2013, at 4:35 AM, A J Stiles asterisk_l...@earthshod.co.uk wrote: Unplug 10.3.22.6, and try pinging it. If something answers, then you indeed have a clash. Check your DHCP server configuration, and make sure any manually-assigned addresses are outside its pool of addresses. If you do this test, remember to make sure to keep pinging with the host disconnected for minimum 30 seconds so as to give your local OS's arp table a chance to time out (or manually delete the original ARP entry before starting the ping). Pete -- Regards, Ahmed Munir Chohan -- _ -- 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] g729 codec over SIP Trunk between CCM and Asterisk
2013/1/17 Onur Cem Çelebi occel...@gmail.com Hello, My problem is, outgoing calls (from asterisk to CCM) work fine but incoming (from CCM to Asterisk) does not work because of CCM is trying to use g729 over SIP trunk. I have found that link after a quick search. Problem is the same as in link below (However my Asterisk version is 1.8.13) and solution seems to have H323 trunk between CCM and Asterisk for using g729 codec. The post was written in 2006. Is there any better solution since that time ? Thanks for reading. link : g279 codec over SIP Trunk between CCM and Asteriskhttps://supportforums.cisco.com/message/1072037 Have you checked if the problem is the license? Asterisk doesn't have a free encoder/decoder for g729, only pass through is available. Try to debug the SIP call to see if the capabilities don't match or just buy a $10 license from Digium (1 concurrent call). Leandro -- _ -- 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] Conf Bridge
Hey all. RE: Conf Bridge. I am looking into a project that would need 8 to 10 thousand parties in a single conference. Most would be on mute but 5 to 6 would be presenters. Is the new conf bridge solid enough to handle this kind of load? Any ideas on hardware projections? If not 8 to 10 thousand how many would be realistic? If not asterisk any other suggestions. Thanks for any input. zktech -- _ -- 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] Conf Bridge
On Thu, Jan 17, 2013 at 3:02 PM, Bryant Zimmerman brya...@zktech.com wrote: Hey all. RE: Conf Bridge. I am looking into a project that would need 8 to 10 thousand parties in a single conference. Most would be on mute but 5 to 6 would be presenters. Is the new conf bridge solid enough to handle this kind of load? Any ideas on hardware projections? If not 8 to 10 thousand how many would be realistic? If not asterisk any other suggestions. Thanks for any input. zktech If most are on mute, then have them call into a stream of the actual conference. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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] Need Help
Hi all! In need of some serious help. We currently run Trixbox and Cent Os on a 2u server for our small business's phones system. We are using some Polycom Soundpoint IP phones. The whole thing came crashing down over the Holidays and as of right now that's about all we have working right now are the phones. The reason I joined this list is because I was hoping to get our external paging intercom system back up and running (it runs off of a sound card but cant get it all configured correctly) and to be honest I have no clue where to start. I've tried reading some online guides but nothing. [MRKlogoblkemail] Joe Ruffolo Director of Operations 801 N State St Unit C Elgin, Il. 60123 847-468-1700v 847-468-0717f j...@mrkgroup.commailto:j...@mrkgroup.com www.mrkgroupltd.comhttp://www.mrkgroupltd.com/ [mrk r2 logo] inline: image001.jpginline: image002.jpg-- _ -- 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] Need Help
On Thu, Jan 17, 2013 at 3:05 PM, Joe Ruffolo j...@mrkgroup.com wrote: Hi all! In need of some serious help. We currently run Trixbox and Cent Os on a 2u server for our small business’s phones system. We are using some Polycom Soundpoint IP phones. The whole thing came crashing down over the Holidays and as of right now that’s about all we have working right now are the phones. The reason I joined this list is because I was hoping to get our external paging intercom system back up and running (it runs off of a sound card but cant get it all configured correctly) and to be honest I have no clue where to start. I’ve tried reading some online guides but nothing. Joe Ruffolo Director of Operations 801 N State St Unit C Elgin, Il. 60123 847-468-1700v 847-468-0717f j...@mrkgroup.com www.mrkgroupltd.com Trixbox forums are here http://fonality.com/trixbox/ as this list will not have the help you might need. -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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] Need Help
On 01/17/2013 09:05 PM, Joe Ruffolo wrote: Hi all! In need of some serious help. We currently run Trixbox and Cent Os on a 2u server for our small business’s phones system. Afaik Trixbox is no longer maintained and their forum are hardly active anymore so it may be a bit of a challenge to get support. If you really need a GUI like Trixbox then I suggest you have a look at Elastix which is very much alive and has a large community and professional services to help you out. See http://www.elastix.org/ Or have a look at Digium's Switchvox (payware). Whatever you do/choose, make sure that your box is secure if you open it up to the Internet. Regards, Patrick -- _ -- 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] Email and web chat call center
Dears; I am using asterisk for call center and I used also VICIDIAL. But they are fine for voice, I need the agents to be able to handle email and web chat messages as long with the voice calls, in addition to be integrated with the social media like Facebook and twitter. Where I can find this? From where I can read and start? Is there a reliable open source solution for this? Regards Bilal-- _ -- 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] Need Help
Hi Joe On 18/01/2013, at 9:05 AM, Joe Ruffolo j...@mrkgroup.com wrote: Hi all! In need of some serious help. We currently run Trixbox and Cent Os on a 2u server for our small business’s phones system. We are using some Polycom Soundpoint IP phones. The whole thing came crashing down over the Holidays and as of right now that’s about all we have working right now are the phones. The reason I joined this list is because I was hoping to get our external paging intercom system back up and running It is useful to distinguish between whether asterisk is working (which this list can help with) or whether its trixbox functionality thats not working When you say phones are working I am assuming you can call in and out and thus asterisk is working fine. The issue is configuration for other parts of the system Trixbox is a lot based around the Freepbx web interface http://www.freepbx.org/ and if its configuration errors their forums or paid support can probably help you However you said it came crashing down, which is concerning as that sounds like hardware issues, generally the platforms are pretty stable unless you get failure or corruption somewhere, often caused by power loss. I would think someone can provide (sell) you remote sys admin / asterisk support to check your hardware, but you probably need to ask for that specifically, or you can buy it via digium or freepbx sites (it runs off of a sound card but cant get it all configured correctly) and to be honest I have no clue where to start. I’ve tried reading some online guides but nothing. My guess is your trixbox paging system sends messages out of your sound card into your physical paging system so two issues you may have are upset configs or hardware failure. Is this your only issue or are other things such as queues and ring groups suffering? Breaking it down into specific pieces will help others understand whether they can help or not I use freepbx a lot but not trixbox or its version of trixbox and alterations, and I am not familiar with sound card setups anymore or Centos (ubuntu is my preference) but there are bound to be others who can help solve specific problems for you It won't be particularly hard, but picking your priority issue and focussing on that is a good first step Good luck Cheers Duncan image001.jpg Joe Ruffolo Director of Operations 801 N State St Unit C Elgin, Il. 60123 847-468-1700v 847-468-0717f j...@mrkgroup.com www.mrkgroupltd.com image002.jpg -- _ -- 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] fw: Re: Conf Bridge
From: Andrew Latham lath...@gmail.com Sent: Thursday, January 17, 2013 3:04 PM To: brya...@zktech.com, Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Conf Bridge On Thu, Jan 17, 2013 at 3:02 PM, Bryant Zimmerman brya...@zktech.com wrote: Hey all. RE: Conf Bridge. I am looking into a project that would need 8 to 10 thousand parties in a single conference. Most would be on mute but 5 to 6 would be presenters. Is the new conf bridge solid enough to handle this kind of load? Any ideas on hardware projections? If not 8 to 10 thousand how many would be realistic? If not asterisk any other suggestions. Thanks for any input. zktech If most are on mute, then have them call into a stream of the actual conference. ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ Andrew sorry for the redundancy please ignore if you like. Andrew Thank you for your feed back. We are looking at all of the options, but how would they call into a stream of the conference? Are you thinking of stacking multiple asterisk servers? The fundamental question I am trying to answer is how many participants could I have in a confbridge and what kind of hardware spec would I need to get the volume. Thanks Bryant -- _ -- 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] Mail list settings?
Hey all For some reason the mailing list is sending all messages from the sending party. This makes it less than ideal when responding; as selecting reply goes to the person and not the list. Can we have it set back to the old way please? Thanks Andrew for pointing this out to me. Bryant -- _ -- 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] Mail list settings?
On Thu, Jan 17, 2013 at 6:32 PM, Bryant Zimmerman brya...@zktech.com wrote: Hey all For some reason the mailing list is sending all messages from the sending party. This makes it less than ideal when responding; as selecting reply goes to the person and not the list. Can we have it set back to the old way please? Thanks Andrew for pointing this out to me. Bryant I just checked back over the list emails and Bryant's email appears to be unique in this problem. I assume it is a simple issue somewhere. List admins? -- ~ Andrew lathama Latham lath...@gmail.com http://lathama.net ~ -- _ -- 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] Mail list settings?
On 18/01/2013, at 12:37 PM, Andrew Latham lath...@gmail.com wrote: On Thu, Jan 17, 2013 at 6:32 PM, Bryant Zimmerman brya...@zktech.com wrote: For some reason the mailing list is sending all messages from the sending party. This makes it less than ideal when responding; as selecting reply goes to the person and not the list. Can we have it set back to the old way please? I just checked back over the list emails and Bryant's email appears to be unique in this problem. I assume it is a simple issue somewhere. List admins? My 2c... Looking back over recent e-mails, it looks to me like Bryant just has a reply-to header on his outbound e-mail's with his e-mail in there. The mailing list is simply allowing his address to remain in the 'reply-to' header (while adding the list's address too). I've noticed some others do this too (Chui Kingh Man) is an example, but there are others. So is this a case of the mailing list no longer stripping 'reply-to' headers before adding it's own, or is this simply a case of a few users setting reply-to when most don't, and those users getting replies directly as well as to the list (as one would expect)? Ie, unless I'm mistaken, it all looks to be operating normally. But I'd be happy to be proven wrong ;) Pete Mundy smime.p7s Description: S/MIME cryptographic signature -- _ -- 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] Mail list settings?
I get direct replies when people reply to my posts. I thought that was just 'cause they wanted to make sure I saw their replies! --Don -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Pete Mundy Sent: Thursday, January 17, 2013 6:57 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Mail list settings? On 18/01/2013, at 12:37 PM, Andrew Latham lath...@gmail.com wrote: On Thu, Jan 17, 2013 at 6:32 PM, Bryant Zimmerman brya...@zktech.com wrote: For some reason the mailing list is sending all messages from the sending party. This makes it less than ideal when responding; as selecting reply goes to the person and not the list. Can we have it set back to the old way please? I just checked back over the list emails and Bryant's email appears to be unique in this problem. I assume it is a simple issue somewhere. List admins? My 2c... Looking back over recent e-mails, it looks to me like Bryant just has a reply-to header on his outbound e-mail's with his e-mail in there. The mailing list is simply allowing his address to remain in the 'reply-to' header (while adding the list's address too). I've noticed some others do this too (Chui Kingh Man) is an example, but there are others. So is this a case of the mailing list no longer stripping 'reply-to' headers before adding it's own, or is this simply a case of a few users setting reply-to when most don't, and those users getting replies directly as well as to the list (as one would expect)? Ie, unless I'm mistaken, it all looks to be operating normally. But I'd be happy to be proven wrong ;) Pete Mundy -- _ -- 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] Open source asterisk GUI options
On 18/01/2013, at 4:28 PM, Jim Boykin boykin...@gmail.com wrote: Hi, We are looking for the web based console for our asterisk system. We came across AsteriskNow but it's kind of bundle and hence not usable for us. What we need is a separate GUI package which we can add to our existing asterisk installs and customize it as needed. Can you help me find what are the Open source asterisk GUI options and how they rates I would go with FreePBX - its very powerful and easy to learn and sits on top of a source or packaged asterisk installation http://www.freepbx.org/ Regards Jim -- _ -- 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] Followme Killing Asterisk
Alright for anyone who ever runs into this in the future, the problem seems
to be resolved by
a) removing the lines Set(Channel(language)=) before the Dial and
possibly
b) using the flags 'dI' with followme app
I guess when using Followme, just don't try and set any another variables
that affect the channel ...after calling it and before the Dial
happens...might work before it.
HTH
\a
On Tue, Jan 15, 2013 at 12:58 PM, A E G all.efor...@gmail.com wrote:
On Tue, Jan 15, 2013 at 11:05 AM, Steve Murphy m...@parsetree.com wrote:
On Mon, Jan 14, 2013 at 9:36 PM, A E G all.efor...@gmail.com wrote:
Hi Guys,
this has been a weekend destroyer for me. I've struggled this all day
and most of today.
From your discussion below, it sounds like the real problem is the
Asterisk crashing.
So, as a first step to solving **that** problem, make sure asterisk is
compiled with debug
flags, dumps another core file, and then you do the gdb asterisk
corefilename, and
get a stack trace. That should give us some idea of what happened.
Thanks for the note Steve. It doesn't sound like there's tremendously
wrong that I'm doing as far s the configuration is concerned then? and it
won't be too surprising since the configuration of Followme is quite simple
assuming the complexities are all handled by the Followme app.
I tried a whole lot of options that made sense as Dial options that the
Local channel dial from Followme is being hooked into but it appears
that, the cause of the crash is most likely that Followme:
1. Is looking for something to do; bill, log or something after it
returns from Dial/call termination but not finding it. I tried using
Answer(nocdr) at the time the call on the DID is being answered but that
didn't help. I have also tried the 'g', 'c', 'C', 'I' and 'i' etc options
with the Dial but they don't help either. I had real hopes in the 'g'
option to tell it to proceed with the dial plan where I was simply making
it return a couple of call status related variables and then just Hangup,
but regardless of the 'calling' or the called party hanging up, these
number get printed, which means that despite the 'g' option, the call does
NOT proceed with the normal/rest of the dialplan
2. Maybe Followme is not built for this purpose where the caller is
unknown (which it would be in most cases) but at least the called party
is usually known AND is a subscriber/registered user of the system who is
then using the Followme feature to find them when they don't answer their
PBX registered phone. What I'm doing calling from outside, having the
system answer the call, allow the caller to put in a number and then
calling those numbers associated with that extension if it's a Followme
extension but the extension itself isn't a registered user in sip.conf or
users.conf, and maybe followme app has some procedures it needs to run
through as a matter of housekeeping (i.e. accounting, billing, logging etc)
that it's not finding info for
Will do a gdb and see what I can find...I'm not a developer so I may not
be able to pick up a lot from the stack-trace but will pastebin it and see
if one of the community/developer members can figure out why it's taking a
dump
Cheers
\a
I have a fairly simple Followme sequence in place to see how it works
before I get into the complex scenarios.
extensions.conf
---
[Incoming]
exten = MyDID,1, Answer()
same = n, Set(CHANNEL(language)=en_AU)
same = n, Followme(TestFollow)
same = n, NoOp(++Back after Followme: DIALSTATUS =
${DIALSTATUS}, Hangupcause = ${HANGUPCAUSE})
same = n, Hangup()
[Followme-Dialout]
exten = _1NXXNXX,1,Set(CHANNEL(language)=en_AU)
same = n, Dial(SIP/GW-1/${EXTEN})
followme.conf
[TestFollow]
context = Followme-Dialout
number = my landline,30
number = my cell phone,20
The call goes out, and rings my first phone. If I answer it, the
Asterisk core dumps, the calls stay up!
snip
[Jan 15 04:19:48] -- Called SIP/GW-1/1203555
[Jan 15 04:19:51] -- SIP/GW-1-0007 is making progress passing it
to Local/1203555@Followme-Dialout-0004;2
[Jan 15 04:19:51] -- Local/1203555@Followme-Dialout-0004;1
is making progress
[Jan 15 04:20:05] -- SIP/GW-1-0007 answered Local/1203555
@Followme-Dialout-0004;2
[Jan 15 04:20:05] -- Local/1203555@Followme-Dialout-0004;1
answered SIP/DIDProvider-1-0006
[Jan 15 04:20:05] -- Starting playback of followme/call-from
[Jan 15 04:20:05] -- Local/1203555@Followme-Dialout-0004;1
Playing 'followme/no-recording.ulaw' (language 'en_AU')
[Jan 15 04:20:05] -- Local/1203555@Followme-Dialout-0004;1
requested a source update
ast00*CLI
Disconnected from Asterisk server
Bus error (core dumped)
...snip
I have been playing with Local channels over
[asterisk-users] Delay in call asterisk
Hi, i am using elastix 2.3 and created some dahdi extensions,now i dialing between the extensions i.e like 2000 to 2001 , but there is delay of 3 to 4 second before it ring the destination. so cany anyone know how fix it so that after dialing the digits the destination should ring . without any delay after dialing. regards Upendra. -- _ -- 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] Delay in call asterisk
If you dial 2001# does it complete the call immediately? Your dial plan may be ambiguous about numbers starting with 2, so it waits a few seconds to see if you're going to dial a longer number. --Don From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of upendra Sent: Thursday, January 17, 2013 11:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Delay in call asterisk Hi, i am using elastix 2.3 and created some dahdi extensions,now i dialing between the extensions i.e like 2000 to 2001 , but there is delay of 3 to 4 second before it ring the destination. so cany anyone know how fix it so that after dialing the digits the destination should ring . without any delay after dialing. regards Upendra. -- _ -- 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] Delay in call asterisk
Hi, yes if i press # then immediately ring , i configured all these by GUI only so how should i fix this issue?? -- Upendra On Fri, Jan 18, 2013 at 11:06 AM, Don Kelly d...@donkelly.biz wrote: If you dial 2001# does it complete the call immediately? ** ** Your dial plan may be ambiguous about numbers starting with “2,” so it waits a few seconds to see if you’re going to dial a longer number. --Don *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *upendra *Sent:* Thursday, January 17, 2013 11:26 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* [asterisk-users] Delay in call asterisk ** ** Hi, ** ** i am using elastix 2.3 and created some dahdi extensions,now i dialing between the extensions i.e like 2000 to 2001 , but there is delay of 3 to 4 second before it ring the destination. so cany anyone know how fix it so that after dialing the digits the destination should ring . without any delay after dialing. ** ** ** ** ** ** regards Upendra. -- _ -- 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] g729 codec over SIP Trunk between CCM and Asterisk
Thanks for reply Leandro. We have installed g279 codec in Asterisk box.Even if not so, there is no problem outgoing (from Asterisk to CCM) calls. But after i searched the issue, i figured out that CCM 4.x does not let g729 codec to pass through over SIP trunk. This is limited only in CCM. If we changed codec g729 into g711u (ulaw) then communication over SIP trunk go on perfectly. Because of CCM does not inject any packets encoded g729 over SIP trunk, i am not able to debug it. But i have tried that i am able to force my SIP phone suscribed Asterisk box to use g729 codec and get work successfully. 2013/1/17 Leandro Dardini ldard...@gmail.com 2013/1/17 Onur Cem Çelebi occel...@gmail.com Hello, My problem is, outgoing calls (from asterisk to CCM) work fine but incoming (from CCM to Asterisk) does not work because of CCM is trying to use g729 over SIP trunk. I have found that link after a quick search. Problem is the same as in link below (However my Asterisk version is 1.8.13) and solution seems to have H323 trunk between CCM and Asterisk for using g729 codec. The post was written in 2006. Is there any better solution since that time ? Thanks for reading. link : g279 codec over SIP Trunk between CCM and Asteriskhttps://supportforums.cisco.com/message/1072037 Have you checked if the problem is the license? Asterisk doesn't have a free encoder/decoder for g729, only pass through is available. Try to debug the SIP call to see if the capabilities don't match or just buy a $10 license from Digium (1 concurrent call). Leandro -- _ -- 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
