similar to: g723.1 voicemail/conference files segfault *

anyone has idea what problem can be here, something with codec but i have today CVS version and grandstream phone with 1.5.0 firmware.I try to change codec in phone and also in asterisk-sip.conf but the same. What can be problem ? tnx, Tomaz *CLI> -- Executing Dial("SIP/102-767c", "CAPI/2:5") in new stack -- Called 2:5 -- CAPI[contr1/2003002]/0 is making

FYI, asterisk DOES now support g723, but you have to pay for it: http://store.yahoo.com/asteriskpbx/asteriskg729.html -----Original Message----- From: Dan Fernandez <danfernandez00@hotmail.com> Date: Mon, 5 May 2003 17:33:05 -0300 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Has anyone gotten SIP_CODEC= to work? Basically, since I?d like to use g723 for sip

Hello, all How can I write sound file with external G723.1 codec ( actually I have CISCO that can make H323 call to Asterisk box with G723.1 or G729 codec ) I am trying to start Record application by specifying in extensions.conf [writesound] exten => s,1, Answer exten => s,2,Record(soundexample:g723sf) or ...... ( soundexample:g729) I'am using oh323 channel driver, in oh323.conf

Hello, I just CVS'd today and now I'm getting these errors when I call one grandstream phone to another both using 711U: NOTICE[1225991360]: File channel.c, Line 1476 (ast_set_read_format): Unable to find a path from ULAW to G723 NOTICE[1225991360]: File channel.c, Line 1446 (ast_set_write_format): Unable to find a path from G723 to ULAW NOTICE[1225991360]: File channel.c, Line 1476

Hello, I have problem to make SIP calls, i have asterisk + PC InterP4 + Digium TDM400P here is the content of the sip.conf: [SIP_PROVIDER] type=peer fromuser=testcomclient username=testcomclient secret=testr host=IP_SIP_PROVIDER ;dtmfmode=rfc2833 context=interne canreinvite=no ;allerid=Beer disallow=all allow=ulaw allow=gsm allow=g723.1 ; Asterisk only

I receive the following error when I try to call another H323 extension from another H323 when going through *. NOTICE[27669]: File channel.c, Line 1325 (ast_set_read_format): Unable to find a path from 1 to 8 NOTICE[27669]: File channel.c, Line 1296 (ast_set_write_format): Unable to find a path from 8 to 1 WARNING[27669]: File chan_h323.c, Line 528 (oh323_write): Asked to transmit frame type 1,

We are having many successes with Asterisk and starting to get the hang of it. But, I am still having problems getting my Budgetone BT100 (firmware 1.0.4.50) to work fully. I can receive calls, but cannot make them. We have the latest version of Asterisk, Fedora Core 3, Digium TDM400P with one FXO and one FXS card configured and working well. We have a PSTN line going into the Digium card,

Good day all I've installed asterisk with sip on my LAN,no special cards,if done sip.conf and extensions.conf and all work 100,I'm using x-lite as a client. I'm trying to do conferencing.What I did was to has out the meetme.conf looks like [rooms] conf => 9876 conf => 2345,9938 and extension.conf exten => 9876,1,MeetMe,9876 When I go onto x-lite and type 9876 it gives me

Licensing info for the G723.1 codec, direct from the holding company that licenses the codec. http://www.dspg.com/technology/LicensePricing.html As you can see they want a LOT of money. This is why I doubt there will ever be G.723.1 codec available fro Asterisk. -- BTEL Consulting 850-484-4535 x2111 (Office) 504-595-3916 x2111 (Experimental) 877-552-0838 (Backup Phone)

Hi All, The meetme.conf have created as below: [rooms] conf => 101 conf => 102 and extensions.conf as below: exten => _1XX,1,MeetMe,${EXTEN} why the warning printed when I called 101. WARNING[27660]: File pbx.c, Line 1051 (pbx_extension_helper): No application 'MeetMe' for extension (ipcentrex, 101, 1) And I found asterisk have not load the meetme.conf when it starts up.

Hello list , I?d like to announce possible problems with migrating any version prior to 1.0.2 to 1.0.3. Pay attention : 1. Codecs Codecs names/description have been changed . For example : versions <= 1.0.2 voip*CLI> show codecs Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. 1 (1 << 0)

Hi all. I've made a patch for chan_oss.c to enable callgroups and pickupgroups in it (since wasn't enabled). I needed it for a special use of the console (pickup calls arriving to the console from another phone) btw, If someone is interested, I can submit a patch to the bugtracker. I won't do it until that's usefult for someone... since is a very special features that probably no

Dear all, I am getting the below errors when using asterisk. I am using sjphone for testing purpose. Below are the setting for sip.conf and extension.conf When i dial the number it rings on the remote telephone. but after ringing 1 time it will disconnect. Can anybody tell me what does this error means and the how to solve this issue. Thanking You, Joel sip.conf [general] context=default

what is g723? ive never seen that before... -- Executing Answer("Zap/1-1", "") in new stack -- Accepting call from '2819870065' to '2815692780' on channel 0/1, span 1 -- Executing Wait("Zap/1-1", "1") in new stack -- Executing SetVar("Zap/1-1", "FAXFILE=/var/spool/asterisk/fax/1105486770.492.tif") in new stack --

I was wondering what are the values for sending dmtf via sip info. I mean, when I use dtmf relay via sip info, the sip/sdp message contains a Signal=X where X is the dmtf. That's ok for dtmf 0-9 . but what when dtmf is * or # ? we must send signal=# ? I ask that because I noticed that budgetones phone sends out * as signal=10 and # as signal=11 . but asterisk don't detect them, 'cause

So, I have submitted my configurations as public samples, and I should have expected this situation to arise. I changed all the relevant "private" configuration data in my samples to obfuscate or alter IP addresses, passwords, etc. However, I left my email address in voicemail.conf... Let me tell you, it took THREE messages sent by a distinctly British-sounding gentleman leaving

Hi, I would like to test chan_sip with a bigger jitter buffer. Does anybody know where in the code this is defined? I looked through it but could not find where. If anybody else can find it please let me know. Regards, Andres

Hi All, I'm currently running Asterisk with SIP phones and an ISDN card using chan_capi. I've just started to use IAX (GSM codec)over the Internet and the sound is adequate. However, there is an occasional 'glitch' in the audio resulting in lost sound or distortion. Is the distortion because I'm using zaprtc for timing instead of a zaptel card, or is more likely to be due to

Hi I have the last firmware for zplex, if you like i send it to you, about the second question 24s means 24 extensions so you can configurate as you wish as fxo or fxs. Att Yelson Vivas

I have a couple of X100P's in my system and while on calls they just randomly hang up for no reason. I have tried messing with the busydetect and callprogress setting them to yes and no same and still random hangups. Is there another setting I should be looking at? My zap config looks like. context = inbound-work include => extensions signalling