similar to: Device state of SIP doesn't change

Greetings, I've noticed a problem that might originate from my Asterisk configuration, could use a hand in sorting it out. Problem is a 488 response from Asterisk whenever it gets RTP/SAVPF profile in the SDP. My current setup has Asterisk Kamailio realtime integration, and Kamailio uses dispatcher to route calls for Asterisk to handle. Now I have only one Asterisk, on the same machine as

Greetings List. we're facing a strange case with my system where in the middle of the call .. after like 7 minutes (not necessarily ) the callee is unable to hear the caller however the caller is able to hear the called party. the scenario is the following. 1- 15 computers running Windows XP SP3 joining a Windows Domain Controller with DHCP , DNS, ISA Internet Acceleration Server. 2- Internet

Greetings listers. I'm running asterisk 1.4.21.2 on SUSE 11.0 using Polycom 501 phones. My outgoing connections are Zapata using a TDM401P. For the most part I can make and receive calls fine except for these 3 issues: 1. When I call an external conference, the call never bridges and hangs up after 60-90 seconds. 2. When I call another number there is a

Hello, I have a problem with a call between 2 webrtc clients. Asterisk removes the ice-related lines from the sdp when it sends the INVITE out, and the called webrtc client rejects the INVITE due to the missing ice lines. Both webrtc clients are defined exactly the same way, same values in all fields except the number of the peer. There's probably something I've changed that causes this

Hello all, I've been testing a Kamailio Asterisk Realtime integration, and found a strange situation. My problem is that when using the integration, everything seems ok but Asterisk does not see the clients as registered. Kamailio and the clients report registered clients. Also calls fail. In Asterisk cli sip show peers shows nothing but for example realtime load sipusers name 660 shows the

Hello. I'm using Asterisk 1.4.22 with Postgres 8.3 in a Ubuntu 8.04 Server. I configured Asterisk to get sip from Postgres, and set qualify for all sips as yes, but the sip show peers command show the status of the peers as UNKNOWN srvcentral*CLI> sip show peers Name/username Host Dyn Nat ACL Port Status Realtime 4900/4900 (Unspecified) D

" Hello, I have been trying to get my Grandstream GXP2000 phones to stop showing ringing state on monitored extensions. But no matter where I put notifyringing=no asterisk still sends the ringing state to the phones. Is this a bug I should report or is there another way around it. Here is how i have my subscriptions setup: extensions.conf [demo] exten => 6100,hint,SIP/100 exten =>

Hello everyone, I'm getting DIALSTATUS=CHANUNAVAIL when a call is trying to get one of my queue interfaces, despite the fact it is free at that time, can you give help? 1. I see many sip channels from that extension: [root at mysweetpbx]# asterisk -rx "*sip show channels*" |grep 648 Peer User/ANR Call ID Seq (Tx/Rx) Format Hold

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

Hello We have recently upgraded our internal PBX from 1.4 to 1.8. This made the BLF lamps on our Polycom phones stop working. After a lot of googling and a lot of testing, I have been unable to find a solution. I did try to change the call-limit value from 4 to 1, and this actually made BLF work (noone suggested this, and what documantation I can find states that this option is deprecated). This

All, I have multiple Asterisk servers in various locations running various 1.4 and 1.6 versions (lab and production) and am having trouble with a new Aastra 6739i (3.0.1.2015) registering. Below is my request to support and they have looked it over and don't see anything wrong: Support, Can not get a 6739i to register with 3 different Asterisk servers with varying configurations/versions

When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. >show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy "Fewest Calls" working for a couple of mouths, and a new agent has been added this

Hello, I'm trying to get calls working between websocket clients and sip clients. For clients I have sip.js based clients on chrome, Zoipers and a Grandstream phone. Challenge here is I'd like to have Kamailio and rtpengine to handle the bridging between different rtp profiles but Asterisk changes them in the sdp bodies along the way. I'm using Asterisk 11.11.0. Is there a way to

Hello, Yesterday night we have upgraded our Asterisk from 1.2.32 to 1.4.24.1 with lipbri 1.4.10, dahdi-linux-2.2.0-rc4 and dahdi-tools-2.2.0-rc2. Libpri and dahdi is only for dahdi dummy cause of the meetme function. After the upgrade we had the problem that some Linksys spa941 phone at one location could not dial out. incoming calls to the phones works without any problem, but outbound the

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

Hi, Does anybody have some ideas - how to play a sound file on channel, after that bridged channel got hanged up? Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. atis at iq-labs.net Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835

Hi, VERBOSE[6120] logger.c: -- Got SIP response 500 "Server Internal Error" I just noticed that i sometimes get those back from provider. They are currently general SIP message log entries with verbose level 3. I wonder if such SIP fails could generate at least WARNING in log? Currently i'm checking logs for warnings and errors, so i probably have missed those.. It would be

Hello Have a setup of asterisk with realtime SIP devices. Trying to organise monitoring of my SIP devices. Once device registered, its state becomes NOT_INUSE (result of DEVICE_STATE(SIP/device) function). Simulating of device breakage - powerdown it. Waiting for a while (minute or two), retrieving DEVICE_STATE(SIP/device) again - no changes! Awaited UNAVAILABLE. doing from CLI: sip qualify peer

Hi, What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. I tried to look at WaitExten and PlayTones, but they seem to not work together - WaitExten doesn't interrupt going on PlayTones. Is there any way how i could do that - so that it looks really natural? It would be silly to create long-long-long

Hi all, i'm trying from several days to setup a SLA on my machine with some THOMSON 2030. My goal is to bind every F key to an extension (NOT a trunk). So, F1 = 201, F2 = 202, F3 = 203, and so on... I'm googled thousand of pages and many more confusing concepts are in my mind. My server uses extensions with numbering 2XX placed in context 'phones'. I set yet in sip.conf: