similar to: meetme music on hold - when only conference member problem

Hi, I have the following problem that when asterisk receives SIP response 302 it cannot forward the call I get such debug: [Jan 4 10:43:27] WARNING[18671]: channel.c:3281 ast_request: No channel type registered for 'Local' [Jan 4 10:43:27] NOTICE[18671]: app_dial.c:505 wait_for_answer: Unable to create local channel for call forward to 'Local/poczta at routing-sip' (cause = 66)

Hi, I am using Debian OS kernel 2.6.22-3-amd64 and zaptel driver 1.4 with ztdummy module for meetme application. I use meetme with SIP channels. I have such problem that when one connects to the conference voice is "cut". Each voice sequence is disturbed. Does any one have similar issue and could give me some advice?? my extension.conf for meetme: ;switch =>

Hi, I have asterisk installed in the xen virtual server. I installed zaptel 1.4.2.1 and patched it to have ztxen module. I loaded ztxen module but when I try to invoke or call to my meetme application I get the following warning and negative result of connecting to conference: [Feb 5 17:46:13] WARNING[10725]: app_meetme.c:772 build_conf: Unable to open pseudo device [Feb 5 17:46:13] --

Hi, I am using Realtime for sip configuration. When there is an INVITE which arrives at asterisk asterisk makes the following selects: Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:138 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM sip_conf WHERE name = 'tzl' [Nov

Hi, I was trying to compiles addons 1.4 and res_config_mysql doesn't compile. is res_config_mysql still supported and is it still posible to use mysql with asterisk RealTime?? Bests Tomasz

Hi, I have the following situation I use asterisk as o gateway between networks. What is the reason for such response? What are the criteria for such evaluation? SIP/2.0 491 Request Pending Via: SIP/2.0/UDP 192.168.129.74:5160;branch=z9hG4bK17c3.17db29e7.0;received= 192.168.129.74 Via: SIP/2.0/UDP 192.168.129.74;branch=z9hG4bK17c3.23083974.0 Via: SIP/2.0/UDP

Date: Mon, 23 Jun 2008 08:00:08 -0400 From: "David Backeberg" <dbackeberg at gmail.com> Subject: Re: [asterisk-users] Replace music-on-hold on MeetMe with ringing sound To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users at lists.digium.com> Message-ID: <3de056a30806230500k7e66185l7bfe473ed398ebf6 at

Hi, I am trying to utilize MWI with sip channel. when my client sens a SUBSCRIBE to asterisk I get info that user not found: <-------------> [Jan 28 11:49:02] --- (19 headers 0 lines) --- [Jan 28 11:49:02] Creating new subscription [Jan 28 11:49:02] Sending to 192.168.129.38 : 7060 (no NAT) [Jan 28 11:49:02] Found peer 'hellboy' [Jan 28 11:49:02] Looking for hellboy in routing-sip

Hi, I have the following situation. I have one account created in my VoIP provider. Asterisk registers this account with the usage of 'register = ' command in the sip.conf file. I have a number of aliases assigned to my user which correspond to different public/PSTN numbers through which I am accessible. When there is an incoming call from my sip provider 'some_extension' which

Hi, Is it possible to filter the calling user with the usage of mysql realtime the same as it is done in extensions.conf file: exten => some_exten/calling-user.... is there some flag which activates this extra check?? Cheers Tomasz

Hi, When I have the following extension matching defined: exten => _an_.,1,NoOp(-- Context routing-sip-announcement for ${EXTEN} --) Asterisk doesn't find it when it receives such SIP request: <--- SIP read from 192.168.129.38:7160 ---> INVITE sip:an_hellboy at ms.sip.rd.touk.pl SIP/2.0 Record-Route: <sip:192.168.129.38:7160;lr=on> ... for instance when I use such extension:

Hi, I saw this problem mentioned before but the user appeared to be using the MP3 software with asterisk. I am using the native music on hold player in asterisk 1.2 and I too have a volume problem with music on hold. Is this controllable through the 'indications.conf'? I know this file controls frequency range for various sounds might it also control sound level or am I barking up

Hi there, I just tried today's CVS: 4/23/2004 version and found a strange loise with music on hold. Basically, when on hold you hear very distorted music as if it was very loud. This is the exact same problem described last year at: http://lists.digium.com/pipermail/asterisk-users/2003-April/009735.html http://lists.digium.com/pipermail/asterisk-users/2003-May/011688.html No answers on

Hello! I've configured Music on Hold in asterisk, the only, most certainly, stupid problem I have is, which DTMFs to send to activate and deactivate it. If I use the cli, I can establish a call with originate. With the "misdn send digit" command I can send a number of digits to the other party. But what are the combinations to put the other one on hold? Or do I have to use a

Hi, what software do i need to convert an mp3 to a g729 format? I have a portal where a user can upload their own MP3, but when a user is using a g729 codec, the music on hold has a crackly sound. using g711 it's very clear. so what i'd like to do is when they upload an MP3 i will make a copy on g729 format, so that asterisk can choose which file to play depending on what codec is

Hello, I've a problem. I've asterisk 1.6.0.5 version. And I've created callcenter, but agents registers to another SIP server. When agent tries transfer a client to another operator , pressing flash, I get this: [Aug 18 16:06:37] WARNING[5259]: chan_sip.c:5349 sip_indicate: Don't know how to indicate condition 9 [Aug 18 16:06:37] WARNING[5259]: channel.c:2858 ast_indicate_data:

For some reason our music on hold is intermittently skipping... running Asterisk 1.6.1.22 anybody know what could be causing this? I don't think it's an encoding problem because it plays fine sometimes. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110502/9e69a5c9/attachment.htm>

I'm running Connected to Asterisk 1.4.18-1. We have a T1 PRI and a SIP trunk coming in from our IVR. When people call in on the SIP trunk, I experience the following problem: I put the caller on hold, MOH starts. MOH quits after approx ten seconds. When I pick the call back up off of hold, they can't hear me for about 20 seconds (sometimes longer or not at all) but I can hear them. This

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi all! During tests with a Grandstream GXP280 phone, I found that pressing 'hold' button, the other extension (Qutecom softphone) is put on hold but without music. Then, when resuming the conversation, I listen the other user again but he/her no longer listen to me. When from softphone the same test is realised, it does not happen this

Hi there, We're experiencing the following problem on a handful of boxes and I'm wondering if anyone has any pointers for me. First up, hardware configs on boxes are the same: Dell 860, Dual Core Xeon, 1Gb RAM, 2x 160Gb SATA's in Software RAID 1. Various Trixbox versions - 2.2.3 and 2.4.2 to name two of them. Problem is that occasionally the music on hold speeds right up -