similar to: Has something changed with AGI recently?

How do I get asterisk to populate the "Calling Party Number" field in an H.323 call? I have asterisk configured to accept a SIP call and connect it to an H.323 IVR system. The call goes through, but the caller id is put in the "Display" field rather than the "Calling Party Number" field. -----Original Message----- From: Skuse, Phil [mailto:Phil.Skuse@vicorp.com]

I'm trying to use H323 for the first time so please forgive me if I've made a mistake here. I have downloaded and compiled the latest versions of pwlib, openh323 and asterisk. I have dtmfmode=inband in h323.conf, but the remote system is not hearing the DTMF. Running a trace reveals the following... 1:08.398 ThreadID=0x00022012 h323.cxx(4594) H323

What's the secret to getting sound through Xlite? The SIP messages all look OK to me, but the sound isn't coming through. It was trying to use GSM, so I searched the archive and tried: disallow=gsm allow=ulaw Now it says that it's using ULAW but I still get no sound in either direction. Phil Skuse (MBJEJPIEUI) <phil.skuse@vicorp.com>

We have an h.323 based IVR platform. When we make a call to it using an h.323 phone, it can see the callers number (ANI), but when we make a call to it via asterisk, the call goes through OK, but we don't get the number. How can I make this work? h323.conf ======= [general] port = 1720 bindaddr = 0.0.0.0 disallow=all allow=alaw dtmfmode=inband [ivr] type=h323 context=default extensions.conf

If I try to create a clearcase view on a Windows 2000 Server share then it creates a subdirectory tree. If I try it on a Samba share, it creates the first directory correctly with the right user mapping and permissions and then returns permission denied. I am able to manually create files and directories on the samba share using explorer, so I guess that clearcase must be using some different SMB

After reading about these $75 SIP phones on this list, I purchased a couple for evaluation. They do work with asterisk - and are good value for money, but as somebody commented: they are not yet perfect. I just wondered if anybody had managed to get either the message-waiting indicator or the conference button to work? Phil Skuse <phil.skuse@vicorp.com>

There's something that I want to set up in our lab for testing purposes, but I'm not sure how to do it. I would like to be able to call an asterisk extension, and then enter a SIP address using DTMF, and then have asterisk make a SIP transfer to that address. For example: If I dial <extn> followed by 4444*192*168*0*10*5060 I would like to be transferred to

Hi, My company wants to put a SIP address on their website. The idea is that potential customers can call that address and will be forwarded to our main switchboard. It's fairly easy in theory because my asterisk server has a real IP address, so any calls to sip:<number>@asterisk-server.mycompany.com should connect just fine (except currently it will be blocked by the firewall). Our

Commit-ID: c36feecd03749ebe7eccb7556c5fc5d38bd88dfd Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=c36feecd03749ebe7eccb7556c5fc5d38bd88dfd Author: Herbert Xu <herbert at gondor.apana.org.au> AuthorDate: Fri, 23 Mar 2018 18:58:47 +0800 Committer: Ben Hutchings <ben at decadent.org.uk> CommitDate: Fri, 25 Jan 2019 02:57:21 +0000 [klibc] expand: Fix ghost

Commit-ID: 887c4118916c8ee1aff8cdefbb691bd835c6566e Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=887c4118916c8ee1aff8cdefbb691bd835c6566e Author: Herbert Xu <herbert at gondor.apana.org.au> AuthorDate: Fri, 23 Mar 2018 18:58:47 +0800 Committer: Ben Hutchings <ben at decadent.org.uk> CommitDate: Sat, 28 Mar 2020 21:42:54 +0000 [klibc] dash: expand: Fix

Commit-ID: af24ffa8f0b9d90e29d6daf77e5349dd3ffe4aec Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=af24ffa8f0b9d90e29d6daf77e5349dd3ffe4aec Author: Herbert Xu <herbert at gondor.apana.org.au> AuthorDate: Wed, 8 Oct 2014 15:24:23 +0800 Committer: Ben Hutchings <ben at decadent.org.uk> CommitDate: Fri, 25 Jan 2019 02:57:21 +0000 [klibc] [EXPAND] Split

Commit-ID: 4dc603d0fe5a997d8bbd5048b229d655e6549ced Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=4dc603d0fe5a997d8bbd5048b229d655e6549ced Author: Herbert Xu <herbert at gondor.apana.org.au> AuthorDate: Wed, 8 Oct 2014 15:24:23 +0800 Committer: Ben Hutchings <ben at decadent.org.uk> CommitDate: Sat, 28 Mar 2020 21:42:54 +0000 [klibc] dash: [EXPAND] Split

Commit-ID: 6e3f1aff00a97256e185897ac25d4d5b9ed5a4a5 Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=6e3f1aff00a97256e185897ac25d4d5b9ed5a4a5 Author: Herbert Xu <herbert at gondor.apana.org.au> AuthorDate: Wed, 8 Oct 2014 20:09:56 +0800 Committer: Ben Hutchings <ben at decadent.org.uk> CommitDate: Fri, 25 Jan 2019 02:57:21 +0000 [klibc] [EXPAND] Optimise

Commit-ID: f253b53e6b2b89b31085e7d1e50f0eb6edf891dd Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=f253b53e6b2b89b31085e7d1e50f0eb6edf891dd Author: Herbert Xu <herbert at gondor.apana.org.au> AuthorDate: Wed, 8 Oct 2014 20:09:56 +0800 Committer: Ben Hutchings <ben at decadent.org.uk> CommitDate: Sat, 28 Mar 2020 21:42:54 +0000 [klibc] dash: [EXPAND]

*Code:* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing Playback("Local/91441425477394 at default-b9f2,1", "sip-silence") in new stack -- Playing 'sip-silence' (language 'en') -- Executing AGI("Local/91441425477394 at default-b9f2,1", "agi:// 127.0.0.1:4577/call_log") in new stack -- AGI Script

*I am getting below CLI in my asterisk :* == Manager 'sendcron' logged off from 127.0.0.1 -- Executing AGI("SIP/cc101-b7910cc0", "agi://127.0.0.1:4577/call_log") in new stack -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0 -- Executing Dial("SIP/cc101-b7910cc0", "SIP/Sama203/119545090201||tTor") in new stack --

Here is another log from the * server CLI, I reall hope some one can help me out on this one. thanks |SELECT count(*) FROM vicidial_auto_calls where status = 'LIVE' and server_ip='127.0.0.1' and campaign_id = '' and call_time < "" and lead_id != '';| -- VDAD get agent: |0|update of vla table: |127.0.0.1 |UPDATE vicidial_live_agents set

My dialer works perfectly , but whenever I dial a number manually from xlite and press a Key like 6055 for DTMF , line gets disconnected. Line gets DC as soon as I press any key from xlite What could be the issues ? I tried the SAME VOIP from another center and Its Ok there. I tried the Same dialer Xlite over Static IP, problem is there. I tried the same number from other Dialer , it works

My Inbound calls lands , but line get disconnect in exactly 3 secs. Here goes my extension.conf setting : [from-ipkall] exten => 901835,1,Ringing ; call ringing exten => 901835,2,Wait(1) ; Wait 1 second for CID delivery from PRI exten => 901835,3,Answer ; Answer the line exten =>

REPOSTED with MORE Info and Modified Subject Line: -------------------------------------------------------- I am using one of the Minute Provider to dial out USA numbers. Now in one of my process, we need to Dial IVR and the enter DTMF digit and then it connects to the automated IVR. When I dial out the IVR directly using Xlite and VOIP Mins provider , it works perfectly. but when In try from