similar to: chan_ooh323 to sip , no connected line info

Hi Gurus, We have configured asterisk to trunk with avaya with ooh323 channel driver. The sip phone registered on asterisk can dial the extensions registered on avaya via this trunk , and vice versa works too. Even we can make the avaya branch to dial asterisk?s extension and then this extension dial back to another avaya?s extension. But if we dial the external DID number via this trunk from

Hello, Users. I loaded module chan_h323.so, chan_vpb.so. I have met a message : "No one is available to answer at this time". I don?t know what I do.. My 'h.323 trace 5' result is : == vpb/1-8: Starting record mode (codec=0)[AST_FORMAT_SLINEAR:VPB_LINEAR] -- Executing Dial("vpb/1-8", "h323/192.168.1.107") in new stack 1:21:34.936 ThreadID=0x06f2bbb0

Hi ..., I've a TE205P card installed in my asterisk box.Port 1 of my card is connected to service provider.From port 2 I want to connect Avay Partner system. what type of cable I require to connect the partner system (straight/cross over). How the call routing from outside will be done to epbx. Thanks inadvance regards rudra -------------- next part -------------- An HTML attachment

>Please I have combed the Archive to no avail on this problem protocol >control problem in oh323. >I'm receiving calls from CISCO AS5300 -> Asterisk -> Zap Channel. The >calls clears the remote location but drops on my own end. Please what >could be >wrong. I have included the oh323.conf and log files. I have tried >various configuration and I thought I should

Have Asterisk connected to a H323 compatible legacy PBX using QSIG protocol and IP trunks. I can call to Asterisk, and from Asterisk using X-Lite softphone but whenever either end picks up, the calls disconnects. No gatekeeper is installed. I have attached a copy of my h323 logfile for debugging. What do you suggest what change needs to take place to keep calls connected? 11:33:19:864

Hi Michael and Everyone I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm getting this error "reason 24 (Call ended with Q.931 cause)" I've checked the Asterisk wiki and several other resources. Please can anyone give me a hint on what the problem is I reach my wits end. Thanks Tola my config and debug Configuration of OpenH323 channel driver

Everyone, I have been trying to connect a PBX with H323 IP trunks with g711 codec to my Asterisk server running ooh323 service. I can place calls to and from either the Asterisk, or PBX with no problem, but when I try to pickup the call on either end, the phone hangs up immediately. Debug shows normal to me but at the last few lines of data there is an error shown that I have not been able to

Hi all, I have VOIP network built with MGCP endpoints.The manufacturer of endpoints is ASKEY. I downloaded latest Asterisk software and found it very useful for me. I configured it and it seems taht everything works OK when I am testing it with one or two endpoints. After that I tried to move Asterisk to working network and replace existing call manager. It starts working and calls are

Greetings All- I have an * box with the NuFone H.323 channel driver installed. I also have an Altigen VoIP system with a PRI to the PSTN. I can sucessfully make a call from a SIP extension (snom190) to an H.323 extension (altigen phone) The thing I can't seem to make work is a call from a H.323 phone to a SIP extension. Here's the layout:

Hi, I'm trying to integrate my asterisk PBX to an Avaya PBX. I am using chan_ooh323 from asterisk-addons. I am able to make a call from SIP Phone -> Asterisk -> Avaya -> Station (phone) and vice versa. I am also able to make a call from SIP Phone -> Asterisk -> Avaya -> PSTN. However I face problems when I make DID calls from the PSTN. The DID calls are made through

Hi all, I would like to call from SIP client to Asterisk then GnuGk, then Cisco 5300 to PSTN phone. Is this possible? I need simple config asterisk and gnugk.Can somebody help me? Ganbaa -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040629/cfe09e1e/attachment.htm

I'm trying to connect an Asterisk 1.6 to an Avaya with gatekeeper (CLAN). When chan_ooh323 first loads it tries to establish a connection with the gk but I it fails. I have the following extract from the ooh323 log. Can anyone give some insight? Thanks! MD 23:02:59:045 Sent GRQ message 23:02:59:045 GkClient Received RAS Message 23:02:59:045 Received RAS Message = { 23:02:59:045

Hi, I recently changed my version of asterisk to 11.XX, and I see a waning with h323, with version 1.8 did not have these warning I have connected one avaya ip office 500 h323 with asterisk and the 1.8 version did not have these messages Oct 23 17:20:35] WARNING[7593][C-000000aa]: chan_ooh323.c:1413 ooh323_indicate: Don't know how to indicate condition 33 on ooh323c_60 [Oct 23 17:20:35]

Hi ! I am trying to connect Asterisk with Avaya Definity. I use this tutorial to do this http://cyril-constantin.blogspot.com/2008/04/howto-connect-avaya-to-asterisk.html The comunication between avaya and asterisk is fine but without sound. I can call from Asterisk to Avaya and extension ring or Avaya to Asterisk and extension ring too but I cant hear anything Example Asterisk ---> Avaya --

We have setup asterisk to handle our calls before between telco and an Avaya definity. The PRI keeps locking up every so often. In addition I keep getting this error when trying to call the avaya: -- Channel 0/2, span 1 got hangup request, cause 102 -- Hungup 'Zap/2-1' When that error happens I get a fast busy (congestion) tone. Any one can point me in the right direction? TIA

05.03.2015 11:42, Dmitry Melekhov ?????: > 05.03.2015 11:29, Dmitry Melekhov ?????: >> Hello! >> >> Just installed asterisk 13.2.0 and see many such messages in log, I >> see them in console during calls, really something like this: >> >> >> -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6", >> "SIP/6166 at

So far I have not found a way that I can register the Avaya phone with Asterisk. From what I have found so far is that Avaya phone needs the Avaya Media Server and Avaya Gateway. Looking at the h.323.conf (in Asterisk) and the file 46xxsettings.txt (avaya file located in tftpboot) there are no settings to make the phone initialize. I have sent an email to the Asterisk Users Mailing List to see

The company I work for has deployed an Avaya IP phone system. They have deployed the Avaya 4602 and 4620 IP telephones. They might be sending me one of these phones for use in my home office. Question: Can I make this IP telephone register and work with my Asterisk server? I don't know if it is a SIP phone? I searched thru the Avaya site, but can't find whether it's a SIP phone or

Hi * Users, I ran into a problem when I was trying to communicate an avaya IP Office talk to asterisk with SIP Trunking. I had successful calls from asterisk to Avaya but not from avaya to asterisk. Can someone provide me insight on how to address it or the path to resolve it. The error I get is mentioned below: (dialing 32564 from avaya to asterisk) "[Nov 6 17:14:23] WARNING[6227]:

Hello! Just installed asterisk 13.2.0 and see many such messages in log, I see them in console during calls, really something like this: -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6", "SIP/6166 at asterisk") in new stack == Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/6166 at asterisk > 0x7fa9d4007660 --