similar to: sip to isdn-capi call problem

Hiya, I sent this bugfix to the asterisk-dev mailing list, and modified it as I noticed side effects, but now it appears to be finished. Nobody seemed to notice it there, so I thought I'd post here, as it seems to be something that will be needed as people update to the latest CVS version. So...read on :) Ted programmer_ted@hotmail.com P.S. Read to the very end. The original bugfix

Hello, I've successfully installed Asterisk from last CVS and configured it for using with DLINK-DG104S as mgcp CPE and PGW2200 as external sip server. All are work fine at G711 codecs, but then I disable all codecs except g729 some calls failed (Not all calls. Some calls passed at g729 succesfully). All my devices configred to use only g729 and I don't see other codecs at mgcp or sip

Dear All, I just setup the Asterisk with E100P which it's no problem in Dial In but I have problem when outdial. The connection method is like this : E1 PRI <-SIGNAL-1-> MaxLink (PBX) <-SIGNAL-2-> E100P <-> Asterisk <--> SIP \-----> Analog PHone Now when I tried to dial out by SIP X-Lite on Windows, it shows me Connect, Trying,

I have a ATA 186 with SIP firmware 3.1 when I changed the configurations to use the g.726 codec I received many erros and the calls doesn't work. I changed the fields: - LBRCodec: 6 <- the code for g.726 - TXCodec: 6 - RxCodec: 6 The errors: Jul 9 13:15:37 NOTICE[1192491824]: rtp.c:500 ast_rtp_read: Unable to calculate samples for format G726 Jul 9 13:15:37 NOTICE[1192491824]:

I have this in sip.conf: [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes progressinband=yes [19256002182] type=friend username=19256002182 callerid="Test hone 1" <+19256002182> host=dynamic canreinvite=no secret=password context=test disallow=all allow=g729 [level3] type=peer host=xxx.yyy.16.99 context=default

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,

I seem to be having a problem with transcoding and/or agreeing on a valid codec. I am running a new image pulled from CVS at 1:30 PM CST. The issue occurs when I try to make a call to a toll-free number over sipphone.com. Here's what I see in the console: NOTICE[1259545280]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[1259545280]: File

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

Whenever I take an inbound call I am getting the following errors: NOTICE[4719]: channel.c:1698 ast_set_write_format: Unable to find a path from speex to gsm NOTICE[4719]: channel.c:1731 ast_set_read_format: Unable to find a path from gsm to speex What typically generates this issue? ~Dan

Hi, Asterisk is new for me. I had a working configuration, but suddenly I can't call anymore with my voip provider. I am not aware that I changed anything in the configuration, but who knows. Can somebody explain me what is happening here? I changed username, password and number. -- Executing Dial("Zap/2-1",

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 i fixed wrong capi.conf (there was [controller1] after [interfaces]) now capi.conf is: ; ; CAPI config ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=855285,859609 incomingmsn=* controller=1,2,3,4 softdtmf=0 accountcode= context=local ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1

hi, all I have test echo application for just fun. I can'nt understand why this is used below in .c file, format = ast_best_codec(chan->nativeformats); ast_set_write_format(chan, format); ast_set_read_format(chan, format); without this this application work fine. then why this is used. any suggestion?? Bhrugu mehta

Hello! I would like to get connected with a VoIP provider for home. At some point, I'm sure I will be connecting to it via an Asterisk box, but for now, I will be using whatever hardware they provide. What recomendations do you in the Asterisk community have for a reliable VoIP service that will hopefully interoperate with Asterisk? A company that is actually willing to work with an

I am experiencing a problem that from list archive it appears others are running into. When I dial from Cisco 7960 via the * to Free World Dialup destinations that supports G.729 the call fails. The major error from the debug log is Jan 15 00:11:14 NOTICE[22545]: channel.c:1481 ast_set_read_format: Unable to find a path from G729A to ULAW Jan 15 00:11:14 NOTICE[22545]: channel.c:1451

I've the following installation : |asterisk client| --- > |asterisk server| --- > |other asterisk server| all the connections are made in IAX, the client and first server allows 711 and 729 the other server only allows 729 since it has low bandwidth at disposal all the numbers but a few are routed to a digium card in the first server, the others are routed to the other server, this

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Hi, here what i have: [2001]--[Asterisk]---[ISDN-Trunk]---[PBX]--[8004] Eicon Diva 4BRI Card to a PBX. Asterisk is running in version 1.0.0 on RedHat Enterprise Linux 3AS with kernel 2.4.21-4.EL. Dialing from Astersik extension 2001 to PBX extension 8004 via ISDN Trunk gives me the following error, -- Executing Dial("SIP/2001-8a8e", "Modem/ttyI0:998004|20|r") in new stack

Hello, I got this information from the manufacturer of my ups Kobian/Mercury and were tested wtih powstatd on Linux (Works OK). Now i want to configure it with NUT and i didn't have any luck (it always says OB) Initiliasation for monitoring: RTS = 1 (High) Main power fail: CTS = 0 (Low) Batter low: DCD = 0 (Low) Kill UPS: DTR = 0 (Low) How would the ups.conf look like? Reagards, Lep

I have come to realize that I don't have to have a g729a license in order to make use of an ATA-186 or 7460 connecting to another 7460. I just need to allow the codec in sip.conf. Now what ramification does that have when I dial out over one of my analog line (connected to * by a channelbank and a T100P) using my 7460 or ATA-186. The only benefit I am looking for is reduced bandwidth

Hi Manfred, I applied the patch and recompiled and reinstalled and I got the folowing warning during my first test call: Aug 19 12:26:51 WARNING[294927]: dsp.c:1234 __ast_dsp_silence: zero length packet It looks like that could be the problem... and the fix! I'll let you know if the problem reoccurs. Might it be an idea to submit the patch to the bugtracker? Thanks, Gary ----- Original