similar to: Asterisk Capabilities

Off topic, but involves an Asterisk deployment in a roundabout way. Anyone here intimately familiar with Cisco Callmanager (Version 4-5), that can tell me where a directory of the standard system voice prompts for Callmanager might be obtained? I am looking for the text and filenames of the standard prompt set that ships with Callmanager, have been all over the Cisco site and I can't find it.

Has anyone had any luck getting a SIP trunk up and working between Callmanager and Asterisk? If so were there any steps you had to take that were not in the documentation on wiki? Blake

Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 answered SIP/MAN-000a4b48 [May 17 00:33:35] DTMF[4238] channel.c:

Hello! Looks like faq, but... Could you , please, point me on how to convert this [cisco] type=friend host=192.168.22.253 insecure=port,invite to pjsip? as you can see another side is very old cisco router, so I can't change anything there. I don't see any examples here

greetings asterisk users :) ive just deployed version 17 and migrated as best I can to pjsip. I can receive calls, and get to my mailbox prompt, however placing calls seems impossible with the following error on dial: Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel (pid = 517890) dunkel*CLI> dunkel*CLI> == Setting global variable 'SIPDOMAIN' to

Callmanager does nothing than construct and tear down calls and the actual RTP stream does not flow through the Callmanager but is direct from IP device to IP device. How does this work with Asterisk? I read something that lead me to believe that Asterisk has to process the entire call, is this the case? Blake Parker CCNA Network Engineer Alacare Home Health & Hospice, Inc. Email:

Hello, I use asterisk to to IAX2 trunking between London POP & Reunion Island pop. I would like to know if it's possible to do a kind of call-limit (i.e. restrict to XX) channels but on a per dialcode and / or destination basis. For example: [trunk] ; reunion proper, i want to send no more than 24 channels exten => _0262XXXXXX,1,Dial(IAX2/mytrunk/${EXTEN}) ; reunion mobile, i want

Hi, I'm facing a strange dialplan issue with a PJSIP_DIAL_CONTACTS. When I try to call an offline endpoint with PJSIP_DIAL_CONTACTS, the dial command breaks and the call control go to hangup block instead of next priority. The error in CLI says "*Dial requires an argument (technology/resource)*". This error seems legit as there are no contacts for an offline endpoint. The dialplan

If understand correctly type=identify is more for sip trunk configuration ? ;[mytrunk] ;type=identify ;endpoint=mytrunk ;match=198.51.100.1 ;match=198.51.100.2 In chan_sip it was just reply 200 OK on keepalive packet without need define trunks. volga629 On Sun, 3 Dec, 2017 at 10:45 AM, Joshua Colp <jcolp at digium.com> wrote: > On Sun, Dec 3, 2017, at 10:42 AM, volga629 at

Hi All, I'm at the end of my tether here and would really appreciate some help. I'm trying to implement DTMF based pause/resume of call recording. I'm using Asterisk 1.4.22.1. Here's the scenario: The caller (SIP or ISDN, doesn't matter) dials into the asterisk which executes the following code: exten => _X.,1,Monitor(wav,${CALLDIR}${UNIQUEID},mb)

How do I configure my DHCP client to restart Shorewall when it obtains a new IP address? Blake

<P>Has anyone managed to get Asterisk to work with Glophone/Voiceglo since this posting.</P> <P><A href="http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html">http://lists.digium.com/pipermail/asterisk-users/2004-February/036559.html</A></P> <P>I've tried copying the config in this listing with no success. </P>

Does anyone have the .bin file(s) to convert a Cisco 7940/7960 back to the CallManager image? I want to start playing around with the chan_skinny addition, but it seems the .exe's from cisco want to open a connection to a SQL server or CallManager (which I don't have).

Hi, ? I am currently trying to get a Cisco Callmanager 4.1 and an Asterisk server (1.6.2.21) to talk via a SIP trunk so I can use the Voicemail component of the Asterisk (all the phones are associated with the Callmanager). The connection seem to be there. When I do a "sip show peers" on the Asterisk server?I see the Callmanager as Monitored and online however I can't get any calls

I have a Cisco Call Manager - and need to use the IVR Feature from Asterisk. My extension is 400 and I am calling 558 on Asterisk In my extension.conf I have these lines : exten => 558,1,Answer exten => 558,2,Playback(message.wav) exten => 558,3,Dial(SIP/439@CallManager) When I call 558 I heared the message then Asterisk tries to call 439 on CallManager but with this error :

Hi Anyone know how to use regcontext et regexten parameter from sip.conf and can give an example ? thx regards Harry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090807/ef9ba45e/attachment.htm

I have experience problems like this in a different scenario. It is usually due to codec translation problem. What is the default codec set on CCM for the IP Phone and the default set in Asterisk? Make sure the defaults are the same. Try G.711 Michael

I have SIP (asterisk 11.23.0) running on port 5060 just fine. udp. I have another SIP trunk thats wants to run on port 5068 (long story). I have enabled tcpenable=yes in sip.conf and defined port=5068 in my trunk definition. It does not seem that anything is listening on 5068? How can I run SIP tcp on port 5068? telnet localhost 5068 Trying 127.0.0.1... telnet: connect to address 127.0.0.1:

Hello everybody, im newbie in VoIP, but find this project asterisk very interesting, i tried to install and its a great sw, i really get sorprised about all of its functions, we need to use the asterisk server in conjunction with cisco callmanager. We have a Cisco Callmanager 4.1 and the clients are softphones from cisco IPCommunicator, but all the support service of our company are linux