Displaying 20 results from an estimated 6000 matches similar to: "Is SIP BROKEN?"
2004 May 14
7
What's in ${EXTEN} ? Why does voicemail prompt for an extension?
Why does voicemail prompt me for an extension instead of just asking my
password?
[voice-mail]
exten => 99,1,VoicemailMain(${EXTEN}@inside)
exten => 99,2,Hangup
Paul Mahler
pmahler@signate.com <mailto:pmahler@signate.com>
<http://www.signate.com/>
Signate, LLC
PO Box 60430
Palo Alto, CA
94306
VoIP Systems, Training & Consulting
2004 Jan 04
3
Newbie - MWI
Sorry for the partial post a moment ago
With help I got two phones communicating - PCMA/PCMU was the problem.
Next stpe is to try voicemail. VM works fine, I can leave a mesage and then
retrieve it - but no MWI on the phone and no stutter dialtone.
I promise I've spent the requisite 4 hours + on google etc. but have really
no further ideas.
The setup is 2 Grandstream phones (latest
2003 Jul 16
8
Call Pickup
Hi,
I have been trying to workout how to use the call pickup.
So Far, I have the following in zapata.conf
[channels]
signalling => fxo_ks
context => local
pickupgroup=1
callgroup=1
channel => 1-3
When I dial *8# all I hear is busy tone.
What have I missed?
thanks
Jay.
2003 Oct 11
1
SIP / IAX over satellite
Hi all,
------
I tried to use * over satellite, but all my effort did not succeed.
The Asterisk is behind the VSAT and is resposibel for alle the SIP
clients in a field location.
The clients are notebooks and PDA's running SJPhoen for Windows and
PocketPC. Unfortunately
I could not find any Linux Client wich worked satisfying. SJ LAbs
promised a Linux Version at the end of
August but they
2005 Jan 27
1
Stumped by BroadVoice SIP
Hello guys.
I am a fairly new user to Asterisk, and I'm just having a tough time.
My goal is to set up a VOIP PBX. I have signed up with a BroadVoice
number, and I have three systems with SIP phones.
The PBX and the SIP phones are all behind a Cisco PIX running NAT.
I am using Asterisk CVS version from yesterday. I also tried 1.0.3 with
little luck.
The SIP phones are two X-Lites on
2004 Aug 15
1
Inbound Free World Dialup - extension not ringing?
Hi to all the * people out there,
Please kind to me as I am both new to Asterisk and to Linux - But I am
learning fast.
My config is quite simple, I'm just following examples and the Wiki: I have
two PC's running X-Lite phones, these work without problems between each
other, and I have a GS BudgeTone-100 registered to Free World Dial UP
(working no problem).
I have tried to
2004 Nov 19
5
Asterisk and H.323 Gatekeeper
Hello,
I am new to this list and to asterisk and going through the archive file I
did not find an answer to my problem.
I have a VoIP network working fine with multiple gateways registered to a
Cisco H.323 Gatekeeper. I have successfully registered Asterisk as a GW in
that network and also successfully registered two X-Lite SIP Client to
asterisk that call to each other.
I want to connect to
2005 Jul 01
1
asterisk newbie and phones which don't want tocomunicate
hi do u have the sip phones extensions in the extension.conf and are they in the right context (sip-incoming)???
are the sip phone registering to asterisk?? try stop asterisk and reconnect as asterisk -vvvvvvvc to check see them registering...
________________________________
From: asterisk-users-bounces@lists.digium.com on behalf of Sistemista WebSolvingJaa
Sent: Fri 7/1/2005 6:43 PM
2003 May 23
12
Unable to create channel of type 'Zap'
I've just installed an X100P, built the kernel module, and tried to use it
to make an outgoing call (via a phone connected to an ATA-186). However, I
just get a reorder tone and see this on the console:
-- Executing Dial("SIP/ata1-4409", "Zap/1/5551212") in new stack
NOTICE[1200825920]: File app_dial.c, Line 481 (dial_exec): Unable to
create channel of type
2004 Apr 07
2
error 488 - Not Acceptable Here
I have a setup of 3 Cisco 7940 running Sip image 6.3. All these phone
are registered by the below information
*CLI> sip show peers
Name/username Host Mask Port Status
2002/2002 192.168.22.199 (D) 255.255.255.255 5060 Unmonitored
2001/2001 192.168.22.200 (D) 255.255.255.255 5060 Unmonitored
2000/2000 192.168.22.198 (D)
2005 Jul 15
2
[Aserisk-Users]no audio inside the net
Hi list, i've problems with my * server and the 4 phones which are
linked to it. i've 2 grandstream bt100 with the firmware upgraded to
101, a wi-fi phone (i don't know its brand) and another ip phone i
don't know its brand. with this sip.conf :
[general]
port = 5060
bindaddr = 192.168.100.229
context = default ;x changed from default to sip
localnet = 192.168.100.0/24
2004 Dec 13
1
Repost: Cisco 7960 and Asterisk...not working....
Anyone help me here? I am a newbie so be gentle ;-)......
It worked once and then I played with the configs.
I have a static IP address which is on my private network.. Phone is 192.192.192.220 and asterisk server is 192.192.192.22
I have the 7690 with a SIP iamge (Whatever latest is )
I have 3 lines setup with Free World Dial up and have the 4th setup to connect to my asterisk server. Here
2003 Jul 11
1
SIP immediate hangups with latest CVS
I've been banging my head on this for several hours, and I have no idea what's going on. Maybe there is a very simple result, and I've been looking too hard at this this evening. This is a brand new system, and I'm wondering if there have been SIP bugs introduced in the latest CVS that are preventing from working what should be a stupendously simple test.
- Cisco 7960
2005 Jul 05
4
Asterisk on Linksys WRT54G
Hi all,
Any one tried installing Asterisk on Linksys WRT54G? We have but facing
problems with SIP to SIP calls. The phones ring and calls are established
but we cannot hear any voice at all. I tried allow=all in the general
section but did not work. So I forced ulaw. Can any one please check it out
and let me know what is wrong?
Here are the conf files:
Asterisk Version: Asterisk
2008 Oct 09
2
Menu for call forwarding or voicemail
I would like to create a simple menu that would allow a caller to
decide whether they want to leave a message or be forwarded to another
number (i.e cell phone). Thanks in advance for any insight.
Here's my current extension.conf
[general]
static=yes
writeprotect=yes
[globals]
[default]
exten => 101,1,Dial(SIP/101,20)
exten => 101,n,Voicemail(101 at default)
;This automatically
2001 Oct 08
1
Hanging ssh session...
Hi All,
I am not sure if this is the same thing as the hang on exit bug, so sorry if
this is a duplication of previous stuff.
Essetntially I am experiencing ssh hangs with about .5% - 1% of my
connections. I am running 2.9p2, on Solaris 7. I actually have empirical
data on the hangings, as I wrote a script to create these connections
in an endless loop, setting an alarm so I could recover
2008 Oct 04
5
Vitelity Asterisk configuration help
I have a Asterisk server setup and I am able to connect to the server
using a soft client 'x-lite' and call and leave a message on my second
extension 102. I have setup a Vitelity account and add what I believe
to be the correct information to my sip.conf and extension.conf. I
would like to setup incoming and outgoing calls with voicemail
support. I've searched all over but many of the
2005 Aug 27
2
Problems with registration
My phone still says Not-Registered. I have a Polycom SoundPoint 600 SIP
phone.
Here is my sip.conf file:
;
; SIP Configuration
;
[general]
context=default ; Default context for incoming calls
port=5060 ;added
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ;
2003 Nov 15
2
Internal server error - cannot align media streams - help needed
Hi,
I configured asterisk on redhat linux 9 box. I installed two different
ip softphones (SJPHONE and X-PRO) and got them registered with asterisk.
The call from one phone to another does get routed via asterisk, but
there is one problem coming up. As soon as call is accepted by the end
user , it is automatically disconnected with the error "cannot align
media streams". If I enable SIP
2003 Dec 18
2
Cisco 7960 - can't traverse NAT?
Might be a stupid question, but is there a default gateway set on the 7960?
-----Original Message-----
From: Paul Mahler [mailto:pmahler@signate.com]
Sent: Thursday, December 18, 2003 7:04 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Cisco 7960 - can't traverse NAT?
I have a 7960 running behind a firewall running NAT. From a telnet session
to the 7960, I can't ping