Displaying 20 results from an estimated 100 matches similar to: "Transfer Calls - REFER"
2006 Mar 27
0
BUG 0003710 - RE: Transfer Calls - REFER
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
2006 Mar 28
2
NATted phones transferring calls - BUG0003710
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
2006 Mar 28
2
Transferring calls - BUG0003710
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
2006 Mar 03
1
Call Transfer - "Both legs must reside on Asterisk box to transfer at this time"
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
2006 Apr 13
2
Asterisk 1.2.7 Page()
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
2006 Mar 19
0
Voicemail Bug?
Ugh.
I have voicemail set up for realtime...
mysql> SELECT * FROM ast_vm_users;
+----------+-------------+-----------+---------+----------+----------+-------+-------+---------------------+
| uniqueid | customer_id | context | mailbox | password | fullname | email | pager | stamp |
2006 Dec 19
0
Is MOH Still Broken in Asterisk 1.4 (beta3)?
I'm wondering if moh is still broken in Asterisk 1.4 beta3. In Asterisk 1.2, when a callee put a caller on hold, the musiconhold class that was played was not the one the callee wanted the caller to hear, but something else. Even after using mohsuggest in Asterisk 1.4, it still appears that this is not working correctly.
Here's the results of a simple test:
CASE CALLER CALLEE
2006 Apr 20
1
Background() and Read()
I'm having some issues with Background() and Read() commands.
See the example below. This is when I wait for Background to finish playing the sound file, before entering '12345#'.
All works fine.
hestia*CLI>
-- Executing Answer("SIP/2944093-3366", "") in new stack
-- Executing Wait("SIP/2944093-3366", "1") in new stack
--
2006 Jun 22
3
Showing Current Calls
Can someone recommend the best way to view current calls in progress on the Asterisk console?
Neither the 'show channels' or 'sip show channels' commands are easy to read.
hestia*CLI> show channels
Channel Location State Application(Data)
SIP/2944093-f9e2 (None) Up Bridged Call(SIP/2944079-e7f2)
SIP/2944079-e7f2
2006 Apr 19
1
Callerid matching in extensions.conf
I'm using Asterisk 1.2.7.1. Has the callerid number matching functionality in extensions.conf changed recently?
exten => 5555,1,NoOp(${CALLERID})
hestia*CLI>
-- Executing NoOp("SIP/2944093-d24d", ""Cletus the Slaw Jawed Yokel" <2944093>") in new stack
== Auto fallthrough, channel 'SIP/2944093-d24d' status is 'UNKNOWN'
This
2006 Mar 17
3
SIP Realtime Users
Trying to get SIP realtime working here...
I'm connected to the database...
*CLI> realtime mysql status
Connected to vox180internal@db1.ipt.XXX.com, port 3306 with username voxadmin for 6 seconds.
I can get information for the extension in question...
*CLI> realtime load sipusers name 2944093
Column Name Column Value
2006 Mar 18
1
Realtime SIP users/peers - Screwed?
Oh heck. It really looks like realtime has been seriously screwed up.
When a call comes in to Asterisk, I can see asterisk executing these queries.
SELECT * FROM ast_sip_peers WHERE host = '2XX.YYY.142.205'
SELECT * FROM ast_sip_peers WHERE name = '2944093'
SELECT * FROM ast_sip_peers WHERE name = '2944093'
So, the first thing it does is check and see if there are any
2006 Mar 22
2
Realtime Query
Arrgh.
I just made a call with Asterisk to extension 2944093. That extension exists in astdb and I have rtcachefriends=yes in sip.conf. Asterisk did a database query...
SELECT * FROM ast_sip_users WHERE name = '2944093'
Uhm... Why?
Doug
2006 May 03
0
Forwarded Numbers and Timeouts
I have a tricky situation. I have a polycom phone with number 3254103. I have configured the phone to forward to a new number, 18059999999.
Here's my dialplan:
exten => 3254103,1,Dial(SIP/3254103,10,tr)
exten => 18059999999,1,Dial(SIP/11101553818059999999@proxy2,40,tr)
When Asterisk dials 3254103, here's what comes up on the console:
hestia*CLI>
-- Executing
2006 Jun 15
7
Executing a Function from AGI
Hmmm. Not having much luck with this. I'm trying to call the DUNDILOOKUP function and assign it to a variable in an AGI script.
I've tried setting with EXEC CMD and with SET VARIABLE. In both cases, it's treating DUNDILOOKUP literally, rather than calling a funciton.
I've tried this:
EXEC "Set" "DIALPATH=${DUNDILOOKUP(2944093|180net)}"
and also:
SET VARIABLE
2006 May 11
4
'extensions reload' clears Regextens
I hope I have this wrong, but when I have a bunch of priority 1 NoOp's created from regexten in sip.conf, and I do an 'extensions reload', I lose all the priority 1 NoOps! This can't be right... this means that in a production environment, if you make a change to your dialplan and do an 'extensions reload', you lose your ability to terminate calls to phones on this system.
2006 Apr 07
2
DIALSTATUS for Multiple Dialled Numbers
Folks,
When I have a dial string like this:
Dial(SIP/3254101&SIP/3254102,20,tr)
and I want to check the ${DIALSTATUS} variable after the dial, how do I know which number I am getting the variable for?
And, what about this?
Dial(SIP/3254101&SIP/3254102@proxy1,20,tr)
What happens in that case? How can I get the ${DIALSTATUS} variable for EACH NUMBER dialled?
Thanks,
Doug.
2006 Nov 29
0
Re: asterisk-users Digest, Vol 28, Issue 152
asterisk-users-request@lists.digium.com wrote:
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2006 Mar 18
0
Realtime SIP users/peers
Just spent hours dicking around with SIP Realtime.
Every time a phone came up and sent a registration to Asterisk, Asterisk would simply NOT query the database. I had sipusers in extconfig, but added sippeers as well. NOW I can see Asterisk doing a 'SELECT * FROM sippeers WHERE name = '2944093''.
Huh??? Uhm, why? It's not a peer! It's a bloody phone, and in my mind
2006 Jun 09
1
Polycom subscriptions
Somewhat off topic.
We upgraded a Polycom phone from SIP v1.6.3 to v1.6.6
The phone will no longer send SIP subscription messages for buddies to Asterisk. I have broken the directory file down to make it as simple as possible.
Here is what it contains.
<?xml version="1.0" encoding="UTF-8" standalone="yes"?>
<!-- $Revision: 1.2 $ $Date: 2004/12/21 18:28:05