similar to: codec pass-through feature

Michael, Thanks a bunch, I downloaded from inaccessnetworks.com thinking that it is the latest :). Ok I will upgrade it. just for the record, following worked. exten => _87.,1,Dial(OH323/H323:${EXTEN:1}@16.52.153.206) Cheers Sathya Date: Fri, 28 Nov 2003 11:28:59 +0200 From: Michael Manousos <manousos@inaccessnetworks.com> Organization: inAccess Networks To:

This appears to have been the magic bullet for me. Thank you very much. So, the bottom line is that there is a bug that ends up making inbound calls use type=peer rather than type=user. Correct? > -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Paul Cheng > Sent: Tuesday, August 10, 2004 8:35

Hi, I am calling from a grandstream phone with g723 codec through * to iconnect. Incoming context as well as outgoing context set to g723.1 codec in *. Call get connected and I can talk. However I get the following warning, which scrolls on my screen until I hang-up. [root@asterisk sath]# cat g723.1 - Executing SetCallerID("SIP/-08122ae0", "1001") in new stack --

Hello, I'm trying to use the GSM codec with Nufone H323 but it's not working. Does somebody has some idea? Have I missed something? Thanks!! Celso Fassoni Some additional info: (I'm using CVS-HEAD - downloaded today) monkey:~# cat /etc/asterisk/h323.conf [general] port = 1720 bindaddr = 192.168.0.100 ; this SHALL contain a single, valid IP address for this machine

Hello, Is there an utility for asterisk for codec conversion? I tried google but i haven' got anything. I am trying to initiate a call with G711 codec to asterisk and i would like asterisk to call a gateway with an g729 codec, therefore making a codec conversion from g711 to g729. I know chan_oh323 does it by specifying the OUT_CODEC variable, but chan_h323 does not. And i was wondering is

I'm attempting to get chan_h323 working on Asterisk CVS-STABLE. I've compiled it ok using the Janus release of pwlib/openh323, by editing the makefile as per the comments. Call setup and cleardown seems to work fine, but no audio is being passed in either direction. Doing an "h.323 trace 9", I noticed the following sequence at the end of the call setup: h323.cxx(1685)

Hi all, we use * as a voip gateway using h323 (chan_h323). We purchased the g729 codec from digium for 10 channels. But, if comes in a call from a cisco gatway or something similar, there're problems with the codecs. If such a call comes in following error occures : Pitch 1st subfr. ! ! Wrong Pitch 1st subfr. ! ! Wrong Pitch 1st subfr. ! ! Wrong Pitch 1st subfr. ! ! Wrong Pitch

Hi, gurus! I'm now using PCM codec in my application and plan to apply AEC to my app. My question is whether I can use AEC module of speex. AEC module of speex can work for PCM codec, too? Thanks, Jinzhe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.xiph.org/pipermail/speex-dev/attachments/20111003/db09011e/attachment.htm

Hi all, Sorry for cross posting. I attached my chan_ooh323 patches (asterisk-addons-1.4.5) when codec negotiation patch changes applied to asterisk-1.4.17. Please let me know whether my patches are correct or not. thanks in advance, Ganbold -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Since the original codec negotiation patch ( http://bugs.digium.com/view.php?id=4825 report) just closed yesterday, and as well as my report (http://bugs.digium.com/view.php?id=11998), I had nothing to do but send my patches to the list. It might be good if my patches are placed at http://www.b2bua.org/wiki/AsteriskCodecNegotiationPatch, but don't know whom should I contact. Anyway

All, Our users have a softphone client that supports the G723 Codec which we want to use for bandwidth reasons, however we do not wish (or have the funds) to license the codec on our Asterisk servers. We have G723 pass-through working between the clients just fine, however calls fail when terminating with Asterisk itself (i.e. Voicemail) or out to the PSTN due to transcoding issues. If it

Hello, I managed to terminate calls from cisco: as5300 and 7206 to asterisk over h323. I tested both oh323 from inaccessnetwork and JerJers chan_h323. I used 1.12.2 version of oh323 and 1.5.2 version of pwlib. After latest changes from JerJer chan_h323.c works ok when receiving traffic from ciscos. I havnt found any audio problems although I didnt send much traffic. Latest oh323 has some

I am using Nu Fone Network's h323 drivers. I can place H323 calls using following in extensions.conf file, exten => _1732.,1,Dial(H323/${EXTEN}@192.168.1.2) If I need to use h323.conf to do the same I cannot configure h323 to do the same. I get everyone is busy message and I do not see IP packets being generated by * trying to communicate to 192.168.1.2. Can someone point out what I

Banging my head against a brick wall trying to get a working H.323 implementation for CVS-HEAD. (The ONLY H.323 I have had working is OH323 v0.6.5 with CVS-STABLE - see my other post regarding compile problems on OH323 for HEAD) So, I thought, lets try this wonderful chan_woomera (dubbed "H.323 for Asterisk that works!"). I get exactly the same kind of problem as I have previously had

Hi Friends, Hope you would help me out here, I have searched the asterisk user list for hours and also read the readme and test files that comes with the driver. I need a very simple scenario. I have SIP clients and want to use oh323 to dial out to PSTN using a h323 gateway. a)If I set the extention.conf like this: exten => _87.,1,Dial(OH323/16.52.153.206) oh323 dials out (I can ring a

Hi list, After several tries to understand the subtil description in the h323.conf to be able to make the next scenario I was presented the following error messages by asterisk. Can somebody tell me please what I am doing wrong. Scenario: Gatekeeper (h323) --> Asterisk PBX -->(h323) Gateway Endpoints are connected to Gatekeeper. Call does come in like 999931235650087 with codec g711

hi We take calls inbound via SIP from our Cisco PSTN gateways, and pass it to customers using IAX (they run their own asterisk servers). We've noticed that asterisk is transcoding the call into a different codec, if the customer prefers a codec different to that which our cisco gw prefers. As such, the quality of the call can degrade. We'd rather asterisk just passed through the RTP

Hi all, I'm trying to trunk my asterisk with a cisco cme 3.3 using h323. Cisco conf: dial-peer voice 8 voip destination-pattern 2... session target ipv4:<asterisk ip> codec g711alaw no vad h323.conf [general] port = 1720 bindaddr = 0.0.0.0 ;tos=lowdelay ; disallow=all allow=alaw allow=ulaw allow=gsm context=from-internal extension.conf [from-internal] exten =>

Greetings, I am progressing well with this great product, the *. SIP to SIP calling, Vocal to *, Voicemail all in the past. Did Iconnect, FWD etc. Also, purchased couple of FXO cards and did zaptel as well. It's time to get to h323 now. Read the mailing list for H323 and OH323 etc. need some help to where to start. Requirement is very simple, SIP calls need to be routed to a third party

Hi folks, I?m using chan_h323 to dial out to a gateway which connects me to the PSTN. In order to use a menu system such my bank menu system, I have to set dtmfmode=info in my sip.conf for my Cisco 7960 phone. However, dtmfmode=info won?t work with Asterisk?s voicemail system. I?m using the g.729 codec for h323 and Asterisk. I?m told dtmfmode=inband won?t work with g.729. Is it possible to use