similar to: Codec problem from SIP to H323

hello, my situation is h323gw - gatekeeper - asterisk - SIP client my problem is, that I can't make call from h323gw, when this GW is Micronet (sp5004). A ----------- CUT ----------- -- Executing Wait("H323/ip$62.152.225.18:52434/20702", "1") in new stack == Spawn extension (postel, 169, 1) exited non-zero on 'H323/ip$62.152.225.18:52434/20702' -----------

Hi i am trying to do the same thing: receive a call from a cisco callmanager and forward it to a SIP user. Asterisk is compiled with h323 support, and is configured as a gateway in the cisco callmanager. h323.conf: [general] port = 1720 bindaddr = 193.x.x.x ; this SHALL contain a single, valid IP address for this machine allow=all extension.conf: exten = 3298,1,Answer exten =

Hi i'm trying to connect to the PSTN in the following way sip ATA -> * -> gnugk -> Cisco AS5300 -> PSTN I'm using asterisk CVS-HEAD-06/01/05-14:33:15 running over RH EL3 Asterisk-Oh323 0.7.2 pre1 Open H323 v1.13.5 pwlib v1.6.6 and I'm having a lot of trouble, gnugk and * both have public ips and are not behind any type of firewall, the sip ATA is behind a firewall and

Hi folks, how could I play an announcement to the calling party as soon, as the called party picked up. I would like to deploy an asterisk in an environment, where a premium rate support-number is offered to customers which do not want to pay a monthly support contract. In Germany, you are commited by law to announce the cost per minute of a premium rate number at the beginning of the call. So,

Hello everyone, One more little problem with a %100 g729 setup. Native moh: musiconhold.conf: [default] mode=files directory=/mnt/kd/moh/default random=yes ; Play the files in a random order ls /mnt/kd/moh/default fpm-calm-river.g729 fpm-calm-river.ulaw fpm-sunshine.g729 fpm-sunshine.ulaw fpm-world-mix.g729 fpm-world-mix.ulaw Place a call on hold: Jun 1 14:55:30

Hi All, I have this codec problem, I use "gsm" in my iax.conf file and in teliax settings also, but the error is still appearing as below. please help me. Kumara Starting simple switch on 'Zap/1-1' -- Executing Dial("Zap/1-1","IAX2/kumara@teliax/01194777070239|30|tr") in new stack -- Called kumara@teliax/01194777070239 -- Call accepted by

Hi Sir, We have integrated Skype with Asterisk (skype user id:- rexesbposolutions). Each call which is coming to skype account is getting transfered to Asterisk Queue. It has following two cases: case 1: When we call from normal skype account to skype account (rexesbposolutions), everything is working fine. case 2: This skype account (rexesbposolutions) has been assigned with a online virtual

Hi list! My problem: I have three extensions in my Asterisk 1.8.30.0 and they have a voicemail. On two of these numbers the voicemail works without any problem, on the other it doesn't... I get this error: [Oct 17 17:01:29] WARNING[14700]: channel.c:5254 set_format: Unable to find a codec translation path from 0x100 (g729) to 0x2 (gsm) [Oct 17 17:01:29] WARNING[14700]: file.c:957

Dears; I am having same problem, that when I place a call from the H323 end point (even if it is not added in the ooh323.conf), then asterisk handle the call and play the wave file in the default context. Also I added endpoint to the ooh323.conf and same thing, it keep goes for default context whatever the context placed. My Asterisk vesion is 1.4.25 My Asterisk add-on version is: 1.4.8 What I

Greetings! to all Now, with some hard time and help from many genurous people's in the list, I have come to this point with my TDM20B card & my teliax's IAX2 account. I hope someone may help me with this issue mentioned below. I have already selected my codec as gms in my iax.conf as well as in teliax's "my account page" but still i have the same error when I attempt

Hi, My configuration is SipPhone<-->*1<--->*2. My asterisk version is 1.4beta3. I installed pwlib,openh323,chan_h323. When i call from SipPhone--(SIP)-->asterisk1---(H323)-->asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Rtp packets are being exchanged. I also tested chan_ooh323, but to fail. Can anyone recommand best

Hi, I have just signed up with Voxee.com and have attached my Asterisk server to dial them via IAX2. Below is the start of the log which dials the number and promply hangs up when the call is answered, with the logs saying that the channel is not compatiable. I have traced this down to the g.729 codec which I don't have installed. Any ideas on how to force that the codec not be used?

Hi, Could someone help me on configuring a H.323 trunk. I am trying to set up the following scenario: [SIPphone(2004)]--[asterisk/oh323/asterisk-oh323]--H323Trunk--[PBX]--[H323phone/(8004)] I am using the following versions: Linux CentOS 3.3/2.4.21-.EL.co asterisk 1.0.1 pwlib_1.5.2 openh323_1.12.2 asterisk-oh323-0.6.3b Calling from Asterisk (2004) to the

set_format: Unable to find a codec translation path from ulaw to g729 Both endpoints are PAP2 set to G711 only I have 1.2.17 on Suse 10.1

I'm sitting here with my dunce cap on. My weak excuse is that I haven't ever played with g729 before. I have a Sipura 841. I have the phone config set to use g729. Its appropriate sip.conf entry, and the IAX stanza for my ITSP all set to disallow=all, allow=g729. But as soon as I dial, I get a complaint from the server: -- Call accepted by 66.225.202.72 (format g729) --

Hi list! I have 4 numbers on my Asterisk 1.8. 3 work perfectly, the 4.th not. I'm not sure, when it finish to work, since a month ago it runs without any problem... Well, if I will be called on this number I can't hear anything and in Asterisk I see these: [Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-000001d1 setting write format

I have build asterisk from latest CVS HEAD-05/09/05 with H323 support as described in README file. Open H.323 version v1.17.1 and PWLib v1.9.0 on Mandrake Linux 10.2 kernel-2.6.11 I tested it with following phones: -- XLite (SIP softphone) -- QMix SIP IP phone (PA168F) -- SJPhone (H323 softphone) -- QMix H323 IP phone (PA168F) -- FireFly (IAX2 softphone) Everything works fine except a problem

Hi, I set a preference order of the codecs to my sip.conf [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls of not registered phones disallow = all allow = g729 allow = g723 allow = alaw allow = ulaw Connected a 'Sipura SPA' sip phone to asterisk with g729 as its preferred codec. Problem: asterisk cannot make

I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only "format=g729" in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the

Hi All, I have set up a box that will be used as follows: SIP Phone ----> Asterisk ----> Cisco H323 VoIP Server 192.168.1.5 192.168.1.50 192.168.1.80 Asterisk is running the latest CVS and oh323 driver. The SIP phone is a Grandstream Budgetone 100. I have everything setup and running with G.711 ALAW and ULAW and i'm able to make calls through