similar to: Transferring calls between two Asterisk Servers

I made the post below earlier today. I'v since removed all NAT from the equation and the problem still persists. Basically I am trying to transfer a call. The transferring phone sends a REFER message to asterisk with a call id that Asterisk doesn't know about. Surely, surely.... someone else must have seen this? hermes*CLI> sip show channels Peer User/ANR Call ID

I made a call from 3254102 to 2944093. I then tried to do a transfer to 3254107. IP addresses have been changed to protect the innocent. It appears this related to bug 3710. It's unclear from the bug if the problem has been fixed or not. If it hasn't, then this seems pretty serious and would I guess affect any NAT-ted phones ability to transfer calls. Here's the REFER that the phone

I have a SIP user, 2944093 that dialled 3254102. I'm trying to transfer the call from 3254102 to 3254104. When I try and transfer the call, I get the following on the Asterisk console. Mar 3 15:14:18 NOTICE[23124]: chan_sip.c:6731 get_refer_info: Supervised transfer requested, but unable to find callid '16749440-c28be02e-64b73be7@172.31.16.67'. Both legs must reside on Asterisk box

I just realised my problem seems to be related to bug 0003710 - "0003710: [patch] Consultative transfers between asterisk servers". It's unclear from the bug info if this problem has been resolved yet. Anyone know? Doug. > -----Original Message----- > From: Douglas Garstang > Sent: Monday, March 27, 2006 4:41 PM > To: 'Asterisk Users Mailing List - Non-Commercial

All, Slightly off topic. Polycom released their SIP software version 1.6.6 for their phones recently. I was under the impression that this release fixed a previous limitation where the phones would only watch 7 buddies, ie send 7 sip subscriptions to Asterisk. I have configured a phone directory to watch 30 or so appearances, and it still seems to only be sending 7 subscriptions to Asterisk.

I have a weird situation. A polycom phone is configured to use system pbx1 as the primary outgoing 'proxy', followed by systems pbx3 and pbx2. All three systems have identical sip.conf files. The phone is registered on pbx1. I shut down the Asterisk application on pbx1. I make a call. The phone sends an INVITE to pbx2. Pbx2 sends back a 407 Proxy Auth message to the phone. The phone sends

Hello All, Ive been doing more and more research on trying to setup a cluster/load balancer for Asterisk. All the Asterisk boxes would be using a config that is the same between them all (via a DB), but we want one location to point the phones to, and from there that machine/device will send it to a Asterisk server so the call can be processed. I know you cant balance the whole call, ie: once the

Has anyone tried to use this? I have: exten => 2944000,1,Queue(some_q) exten => 2944000,2,Set(QUEUE_PRIO=10) exten => 2944000,3,Queue(some_q) When the user enters the queue again, they are being put at the back of the queue. It seems this new variable does not work. Doug.

Is it possible to have nested MySQL queries in extensions.conf? Ie, perform a query, grab a value, and then jump to another location in the dialplan and do another query based on that original value. I'm having problems with the result and fetchid's and I'm not sure if it's even possible to do this or not. Thanks, Doug. -------------- next part -------------- An HTML

I made a call from 3254102 to 2944093. I then tried to do a transfer to 3254107. IP addresses have been changed to protect the innocent. Here's the REFER that the phone at 2944093 sends directly to Asterisk: U 216.186.128.68:5060 -> 216.186.142.203:5060 REFER sip:3254102@216.186.142.203 SIP/2.0. Via: SIP/2.0/UDP 216.186.128.68;branch=z9hG4bKba3b074892377BD1. From:

Here's an interesting question: If I transfer a call from Asterisk system to another with IAX, is there any way I can get control back on the original system? Or.. do I lose control, and the dialplan has to continue on the new system? Scenario is we transfer calls to an Asterisk system that handles ACD queues. If the ACD queue times out, we want to send the caller to voicemail on another

I just upgraded to Asterisk 1.2.7 from 1.2.5. Page() is behaving differently. I'm getting an error - Incomplete destination '' supplied. -- Executing Page("SIP/2944093-5999", "SIP/3254107&SIP/3254105|") in new stack Apr 13 11:06:11 WARNING[9294]: app_page.c:193 page_exec: Incomplete destination '' supplied. -- Playing 'beep' (language

Is there any way to get Asterisk to reload the /var/lib/asterisk/astdb file? It seems to only read it on startup. Thanks.

Greetings all- I've got an odd issue with a QuickPhones QA-342 WiFi SIP phone. It registers correctly, makes calls, etc with no problems. The dtmfmode is set to rfc2833. HOWEVER, I'm unable to transfer calls with it. The proper procedure should be to hit the 'Call' key which sends the flash event, we should get dial tone, dial the number to transfer to, then hit 'Call'

Hi everyone, I have a question concerning DNS SRV lookups. The situation is like this: - one central Asterisk server - many domains with SRV records, let's say we have bar.com and doe.com Now the question is: if the SRV lookup is done for foo@bar.com the call is mapped to foo@myasterisk.mydomain.net. Is that correct? If so, I have a problem: if somebody calls foo@bar.com, Asterisk

I have the following in sip.conf. It was my understanding that this configuration (ie with deny/permit) would only allow connections from hosts 192.168.10.4 and 192.168.10.5. That doesn't seem to be the case. Asterisk is accepting INVITE's from other addresses. [a00090101] type=friend context=Company1 username=a00090101 ;secret=180 ;insecure=very host=dynamic mailbox=company1@vmusers

Hi all, I am running Asterisk in conjunction with a Sip proxy. Asterisk is registered to an external SIP carrier (sip.uni.it) If a call reachs Asterisk through the SIP carrier, then it is forwarded to the external SIP proxy extension (530 at weboffice.dyndns.org), when the extension 530 that has answered the call tries to transfer the call to another extension (513 at

Anyone know if #include works in ael yet? extensions.ael: #include "inc/pbx/global.conf" context test_context { }; *CLI> ael reload May 30 13:56:45 NOTICE[8516]: pbx_ael.c:1120 handle_root_token: Unknown root token '#include' May 30 13:56:45 WARNING[8516]: pbx.c:3758 ast_merge_contexts_and_delete: Requested contexts didn't get merged

Wondering if anyone on here can help with a niggling issue: One of our extensions is unable to make attended transfers at all. The phone in question is an Elmeg ip290, and works fine for direct transfers. However, on attempting to make an attended transfer, the first leg succeeds (the inbound call is placed on hold and gets MoH, the Elmeg user announces the call to the target extension), but upon

I'm wondering if the page application is broken in 1.2.5 The following: exten => 2001,1,Page(SIP/3254105) does strange stuff. The caller's phone immediately drops into the call, while the callee's phone is still ringing. I'd think it was a SIP messaging issue, except that the Dial() command is working fine, which makes me wonder if it's a bug in the Page appplication.