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Does anyone know how I can get stderr from AGI to be sent to somewhere other than the console? It seems that this is the only place it can go. Changing logger.conf has no effect. If you want to see errors from AGI scripts, you have to run the Asterisk console, which isn't viable. Doug.

Hello, I am just asking this because I am note sure if the problem is on my side or not, I saw some comments on SIP realtime today so I was wondering, has anybody has SIP realtime working with a softfone ? If yes, please confirm, that would give me a light. My previous message to the list is below. Thanks. Frederic ----- Original Message ----- From: Frederic Jean To:

Hello everybody, I have this problem where I can't get a ring tone when SIP devices dial an IAX2 route. I get the ring tone using IAX2 devices to dial the same route. The call completes normally in both cases... Facts: - Asterisk 1.0.9 - The Dial command is within an AGI. - ATA (grandstream) and firefly (SIP mode) would not give me the ring tone at all - Switching to a SIP route works ok -

Hi everybody, I sent an e-mail this morning regarding SIP / IAX2 with no ring-back, I now succeeded to pin-point the problem, here it is, if I dial a provider directly from extensions.conf I get ring-back, if I dial from an AGI script I don't get the ring-back but it calls anyway. I use 1.0.9. Any hint would be appreciated ! Thanks, Frederic ;Calling this one does not give me ring back

Hello, I upgraded from 1.0.9 to 1.2.5 yesterday and it went ok except for one little change that happened in the CDRs, and it's concerning the CallerID. Let me explain.. With 1.0.9 I used to get this in the CDRs: "COMPANY LTDA <2153>" Now with 1.2.5 I get one part only: "2153" The number 2153 in this case is the username, but where is the text part gone ? that

Hi, Can someone explain to me how to set up the sip_buddies table from 1.2.5 properly so my users can authenticate correctly without using the "name" field ? (if it's possible) First I was assuming that it would be possible for a user to connect and dial just providing "username","secret","host" and "context" but it seems that I need

Hello all, Is there a way to check who is using the VAD option whenever we get the message "VAD frame at the end...." at the CLI ? --- The IP is not listed. Some people use it but I can't tell them to turn it off for better audio performance, and I know it generates a lot of messages on the system as well Thanks in advance Frederic

Hi everybody, I am almost there on that one :-) Transfering a DID from SER to Asterisk 1.2.6, but I get 403 forbidden. I tried this example but without success and I also looked at last year's posts.. http://www.cs.colostate.edu/~somlo/CSU-SIP-notes/#3.3 SER is the public access and is on a separate box. The URI is sip:551130256898@sip.provider.com and the Asterisk is correctly

Hi, downloaded the latest snapshot just right now at soft-switch for MFC/R2 support and I get this message when trying to do libmfcr2; any idea ? it looks like he's not reaching Unicall. Thanks, Fred # make make all-am make[1]: Entering directory `/usr/src/libmfcr2-0.0.3' if /bin/sh ./libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I. -I. -I/usr/include/libxml2 -g -O2 -MT

Hello, I installed 1.2.5 and realtime SIP. The connection to the DB is OK because I can get the values from the CLI. Here are my 3 different cases: 1- If I put an unexisting user, I get 404 and I am not able to dial. 2- If I check "Disable registration" within Firefly it does not register but I am able to dial a destination (...) 3- If I leave registration ON, I get the 404 message

Good morning everybody, Can someone explain to me the interconnection between these four things: indications.conf, SetLanguage(), zaptel.conf and ring-back ? If there is any !! :- ) I am having this case where some users cannot hear ring back from a DeadAGI script and it seems to be interconnected to these items. These users are from the iaxfriends table, they _can_ hear ring-back from a