Displaying 20 results from an estimated 2000 matches similar to: "Re: [asterisk-dev] bug or bad chan_sip.c"
2006 Apr 08
0
Re: [asterisk-dev] bug or bad chan_sip.c
Tzafrir,
How did you set sip:tzafrir@local.xorcom.com
I use ser----asterisk
look at my sip.conf and extensions.conf
Regards
Harry
////////////////////////////////////////////////////
[general]
context=sip
realm=nxs.yi.org
bindport=5050
bindaddr=nxs.yi.org
srvlookup=yes
tos=lowdelay
maxexpirey=3600
defaultexpirey=1000
allow=all
musicclass=default
language=fr
insecure=very
allowguest=yes
2006 Mar 30
0
Wrong extension indicated when logging in as agent
Hi,
I am not sure if this is a bug in FOP (Flash Operator Panel), a
configuration error on my part or a bug in Asterisk. I am using Asterisk
1.2.5 and Zaptel 1.2.4 on a Centos 4.2 server with Linux version
2.6.9-22-EL-i686. Kernel updates are excluded and the server has been
updated using 'yum -y update'. Okay here is the scenario: I am using
AgentCallBackLogin as an extension in my
2006 Apr 10
1
Call me for testing my system
Dear User,
Anybody could dial these sip uri :
sip:info@nxs.yi.org (french)
sip:music@nxs.yi.org (music 60s)
sip:support@nxs.yi.org (french)
Thanks for help
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2005 Sep 21
1
Call getting disconnected in queue
Hi,
I have a small call center with 4 Zap lines and 4 agents. Agents login
using sip phones with AgentCallbackLogin. I occasionally gets a
complaint that when customers call the call center, after the initial
greeting is over the call gets cut after playing the thank you message.
I started investigating and found that that happens when the call gets
transferred to an agent who is making an
2006 Apr 28
1
[SPAM] [asterisk-dev] Disable 407 proxy authentication for outbound domains
Hello,
I posted a lot of mails may be asterisk is not able to
accept sip calls from internet !?
My english is not fluent i try my best !
My problem I use ser+asterisk.
For local calls there are no problem (PSTN or IP)
Now i wish to receive calls from other internet domain
but asterisk ask for authentication 407.
IS IT possible to Disable authentication for incoming
calls to my sip uri ?
2005 Jul 23
2
(cause 66 - Channel not implemented) -- IAX?
Hi,
I am setting up a small call center using *. I have ZAP setup for
incoming calls and IAX setup for agents. Agents login using
AgentCallbackLogin. When customers call, it's getting picked up and when
queue is trying to call back the agents, I am getting error.
I am using CVS HEAD, and updated just now.
The error is:
-- Executing Answer("Zap/1-1", "") in new
2006 Feb 13
1
Asterisk: Agent logs into queue, and there are calls in the queue, but calls don't go to agent
Here is some dialog from the Console:
-- Starting simple switch on 'Zap/13-1'
Feb 10 07:22:36 NOTICE[21105]: chan_zap.c:6063 ss_thread: Got event 18 (Ring
Begin)...
-- Executing Goto("Zap/13-1", "mainmenu|s|1") in new stack
-- Goto (mainmenu,s,1)
-- Executing BackGround("Zap/13-1", "thank-you-for-calling-poker -support")
in new stack
2004 Sep 08
4
Cisco GW and DTMF problems
I'm stuck running a old copy of asterisk because something strange is going
on in later versions of the CVS..
When I call in from a PSTN into my cisco 2610XM gateway which then routes
the call to my asterisk box via sip.. Asterisk can no longer process DTMF
tones generated by the calling party. This affects DISA, prompts and
menus.. Has anyone else had this problem?? and use.. I DO have
2006 Mar 06
0
No ring when doing blind transfer.
Hi,
I have an odd problem when doing a blind transfer. The transfer is
intiated and the transferred caller hears nothing until the timeout. I
have tried setting the 'r' and the 'm' variables in the dial command.
Nothing happens when I use the 'r' variable when I use the 'm' variable
I briefly hear music on hold and then it stops until the timeout for no
answer
2004 Jan 26
0
canreinvite and codec negotations... and NAT
I've gotten canreinvite=yes to work with a sip device behind NAT!! You
*MUST* port forward the SIPPort to in your gateway router to your phone.
This is a MUST.
Okay, now on to my problem.. I have people who will be using ulaw, and I
have people who will be using g729.. I want to set it up so that canreinivte
will work.. I have a single cisco gateway..
Asterisks isn't handling the
2004 Jan 29
0
canreinvite and codec negotations...
Okay, now on to my problem.. I have people who will be using ulaw, and I
have people who will be using g729.. I want to set it up so that canreinivte
will work.. I have a single cisco gateway..
Asterisks isn't handling the negotation between the 2 devices very well..
For example..
[gateway]
type=friend
host=1.2.3.4
canreinvite=yes
qualify=200
dtmfmode=rfc2833
context=default
disallow=all
2004 Apr 13
0
Bug with 'r' in dial
The lastest CVS's versions (both stable and head), the 'r' option in
app_dial doesn't work with SIP and Re-invites. I've heard reports that it's
not working with IAX2 either.. I'm using Cisco gateway and cisco ATA's and
I am doing re-invites, and it's worked up till this point.. What's going on?
Thanks, Billy
2004 Apr 27
0
Strange Warnings and dropped sip calls.
I've been getting this Warning message for a while now..
Apr 27 13:56:45 WARNING[1142106560]: chan_sip.c:5775 sipsock_read: Recv
error: Resource temporarily unavailable
and from what I can tell, this warning coinsides with a dropped call..
I'm running Cisco Gateways with Cisco ATA's (running 3.1 firmware) and I am
doing Re-invites with NAT & STUN (and in some cases RTP aware
2007 Aug 26
0
Nokia cell connectel to asterisk
I use the E-series Nokia phones on my Wireless LAN. The e series have sip agent
On 8/20/07, asterisk-users-request at lists.digium.com
<asterisk-users-request at lists.digium.com> wrote:
> Send asterisk-users mailing list submissions to
> asterisk-users at lists.digium.com
>
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>
2006 Feb 22
2
context being ignored by inbound sip call
hello-
i was messing around with a did from ipkall.com, and asterisk seems
to be ignoring the context specified in the sip config.
in sip.conf, i've added:
[7508] ;ipkall
type = peer
dtmfmode = rfc2833
context = remote
callerid = "ipkall incoming" <7508>
nat = no
in extensions,conf, i have:
[remote]
exten => 7508,1,DISA(1111|internal)
[internal]
exten =>
2003 Oct 31
1
Problems with SIP
I'm new to Asterisk, but, Managed to get it working for outound calls from
my ATA --> Asterisk --> Cisco 2620 using SIP. However, I'm having problems
with Inbound calls from the Cisco.. Cisco 2620 --> Asterisk --> ATA .. In
fact, voice mail won't even work..
This is a snippet of what I'm getting when I try to call the ATA
-- Executing
2006 Mar 30
0
BUG: FOP reports incorrect (duplicate) IP address until restarted
Hi,
This problem may be related to a configuration problem but I believe it
is a bug in the FOP since restarting the FOP server clears the problem.
Here is the scenario: Using AgentCallBackLogin and have four agents
logged in a call is made to one of the agents directly from an internal
phone. Okay so far. Call is hung up and the same extension is used to
call another agent okay again, no
2007 Jul 12
0
No subject
such file or directory" on pure-IP platform in which I installed
asterisk-libpri-dahdi trilogy.
Maybe, it's me while following README instructions, maybe README
instructions could be improved or maybe it's wrongly labeled messages ?
That's why I told myself : I'm waiting too much from doc ? is a pure-IP
platform too specific ? what is the official policy ?
README starts with
2006 Aug 11
2
AgentcallbackLogin()
Can someone tell me why this is not valid...
[start]
exten => 1000,1,Answer
exten => 1000,2,Wait,1
exten => 1000,3,AgentcallbackLogin(1000||2000@Local)
exten => 2000,1,Macro(DialProxy,115551212)
exten => 3000,1,Queue(testq||||45)
while this is:
[start]
exten => 1000,1,Answer
exten => 1000,2,Wait,1
exten => 1000,3,AgentcallbackLogin(1000||2000@start)
exten =>
2005 Feb 02
0
RES: AgentLogin / AgentCallbackLogin transfer pro blem
Hmm i found the problem... I?m using a Grandstream BT100. The transfer just
works in a queue if I first acknowledged the call using the # key, and then
press the TRANSFER key in the Grandstream.
In the asterisk console I receive a:
-- SIP/4002-4563 acknowledged
Then I can transfer the call... Weird because i?m using ackcall=NO in
agents.conf ...
Diego Magalh?es
diego@redetaho.com.br
+55 24