Displaying 20 results from an estimated 1000 matches similar to: "forward call back up same trunk to external cell phone problem"
2005 Jan 13
1
Registration of SIP
Hi,
I am getting this problem when trying to register with Voipfone.co.uk. It
used to work, and I haven't changed anything that I know of.
Jan 13 10:22:37 WARNING[21645]: acl.c:213 ast_get_ip_or_srv: Unable to
lookup 'voipfone.co.uk.voipfone.co.uk'
Why does the domain name appear twice? I don't know when it stopped working.
In SIP.CONF
[sip_proxy-out]
type=peer
2010 May 28
2
Asterisk 1.6.2.7 + app_fax + OpenBSD 4.7 minor issue
Hi folks,
I am having a small problem with asterisk-1.6.2.7 + app_fax on OpenBSD
4.7 -release. Everything seems to work fine. I have a macro which
answers, receives the fax to a tiff, and then runs a script (mailfax) to
convert that to pdf and email it. It all works perfectly except for some
errors I am seeing in the console. After it hangs up I get a dozen or so
messages in the cli
2011 Nov 21
1
video calls not working
Hi list,*
*I am not able to make video calls between two sip accounts. below is the
information. please help me where I am missing the configuration.*
Extensions.conf*
exten => 111,1,Answer()
same => n,Dial(SIP/2206,60,r)
same => n,Hangup()
*SIP.conf*
[2218]
type=friend
secret=*******
callerid="Virendra" <9172341457>
host=dynamic ;
2007 Jun 26
1
Asterisk to Cisco 2600 GW DTMF Not Working
Hi All,
I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router
with a PRI card in it, handing off to a PBX and vise verse. Calls in
and out are working fine except for DTMF from Asterisk to the 2600.
DTMF from the 2600 to Asterisk is fine.
Here are the Asterisk console warnings I get when I send DTMF from
Asterisk to the 2600:
== Forcing Marker bit, because SSRC has changed
Jun
2004 Nov 23
1
Fax over SIP Problems (sorry for this topic ...)
Hello everyone!
I tried to send a fax over SIP with an Asterisk Server in the middle (no
Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway
is external). Whenever I start sending a Fax to a PSTN destination, the
Call gets answered and asterisk tries to build a native bridging:
-- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and
SIP/xxx-3ef8
Then the following
2005 Jul 11
4
Video phone settings???
I have three video phones here for testing:
Extension 6003 is Eyebeam
Extension 6004 is a hard phone (model 8770)
Extension 6005 is a hard phone (model 8882)
Can anybody have a look at my settings and the output I get from all
kinds of dialings, please.
The sip settings for all phones is (user / password different):
[6003]
type=friend
username=6003
secret=pwd
qualify=200
nat=yes
host=dynamic
2017 Apr 19
2
Can't compile Asterisk on Ubuntu 16
Hey;
Thank you very much. I was able to install asterisk from your link. One
other question. How are you starting asterisk? Do you use an init script or
systemd? Do you think that you could share the script you use?
Thanks Again;
John V.
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Jonathan H
Sent:
2009 Jul 16
3
T38 negotiation, the last step !
Hi, I've managed to get HYLAFAX---->T38MODEM----->ASTERISK---->CISCOAS5400
working, but when they are negotiating asterisk drops a message telling
"Unknown RTP codec 96 received from gateway" Do somebody know how to fix it
?
Thank you !
<< [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [SIP/GWCISCO5400O-600bfcc8]
<< [ TYPE: Control (4) SUBCLASS: Answer (4) ]
2004 May 05
1
Asterisk devel. - Mediatrix dtmf bug solved
Hello,
When using Asterisk version 0.7.2, FreeBSD port with Mediatrix 1124 gateway,
there is problem with DTMF "out-of-band".
See debug below: Mediatrix forces (*) to use Payload Type as 96:
[...]
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
[...]
Then we've got this nice debug from (*):
May
2003 Jul 31
1
RTP codec 13 received - Cisco incompatibility?
Hello,
In our SIP network, Asterisk is the central PBX, and it routes calls to the
PSTN thru a Cisco Router - IOS 12.2(11)T9.
If a client softphone calls directly via Cisco to the PSTN, the call works
successfully.
If the client softphone calls via Asterisk to other SIP internal extension,
it work fine too.
The problem is when a client calls an Asterisk extension, and Asterisk
transfers
2003 Jul 31
1
RTP codec 13 received - Cisco incompatibilit y?
I have a similar setup to you and get the same message regularly. I don't
think it's the cause of your problem. I did some research on it a while ago:
IIRC the cisco uses codec 13 for "silence suppression" whereas asterisk
(correctly) uses codec 19. The router can be configured to use 19 also, but
I didn't bother. I'm sure somebody will correct me if I'm wrong about
2016 May 16
2
Asterisk 11 on Centos: Voicemail crashes when recording message
Hi folks,
I'm running Asterisk 11 (at the moment - planning to u/grade to v13.7
LTS), I've just configured the voicemail function, and it's mostly
working fine... except when I try to leave a voicemail! This crashes
asterisk with no entries in the messages log.
The system is running on Centos 6 (or maybe 6.5, I'm not sure how to
check this). uname -a returns:
Linux
2004 May 12
1
Musical interruptions
Whilst on a call, I'm getting the following...
-- Started music on hold, class 'default', on SIP/phone3-a7d5
-- Playing 'pbx-transfer' (language 'en')
-- Unable to find extension '#' in context 'default'
-- Playing 'pbx-invalid' (language 'en')
ie - without anyone pushing keys - I hear the music on Hold - as does
the
2006 Oct 31
0
SIP with Qualify and NAT
Hi guys,
I'm having a really strange problem, which I'm pretty sure has only
appeared since my last upgrade (1.2.12.1) .
It's about NAT and Qualify. I'm using Asterisk to register with some
external SIP providers. However, they're always marked as UNREACHABLE,
when they weren't before!
A typical debug looks like this:
hera*CLI> sip reload
Reloading
2004 Jul 27
2
Using rxfax over SIP
I have no analog line interfaces on my asterisk system, but I do have two UK
0870 numbers routed to two separate VoIP accounts (one with FWD, one with
gossiptel). Asterisk is configured to register with these accounts. I get
voice calls through just fine this way.
I thought I could get one of these 0870 numbers to route through to rxfax,
thus allowing folks to fax me directly.
I've set up
2008 Nov 27
0
trunk peer not registering after migrating installation
I have an odd problem. I have just installed asterisk on an ubuntu
box, and migrated the previous configuration of asterisk (on another
ubuntu box) to this new server (scp -pr xxx at oldserver:/etc/asterisk/*
/etc/asterisk/)
Asterisk worked fine on the old server, but on this server my SIP
trunk peer does not login after initial server startup. "sip show
peers" shows my phones
2007 Dec 28
0
call queuing not detecting caller hang up when call originates from voip provider
Dear all
I've got call queuing working when calls originate from my local site.
After testing I migrated it to calls originating from our voip
provider- it should ring an extension, then queue . All works well
apart from if the caller hangs up when queued: the call hangs around
in the queue as a phantom until one of the extensions answers it and
it is destroyed
Am I doing something wrong?
2005 Jun 27
2
is teliax down?
I'm getting really wierd errors from them, like bad packet checksums:
Jun 27 18:22:56 NOTICE[29051]: rtp.c:435 ast_rtp_read: RTP: Received packet with bad UDP checksum
- a
--
"I didn't see it then, but it turned out that getting fired was the
best thing that could have ever happened to me. The heaviness of
being successful was replaced by the lightness of being a beginner
2005 Jul 27
2
"Received packet with bad UDP checksum" - whats the real problem?
We have a customer trying to dial through our server, and our server is
throwing tons of these log messages:
Jul 27 14:21:02 NOTICE[29210]: rtp.c:431 ast_rtp_read: RTP: Received
packet with bad UDP checksum
Is it pretty certain, that these are caused by a bad or misconfigured
router along the path, or something else network-related? As opposed to
the SIP hardware itself? The SIP ATA is the same
2005 Sep 27
0
asterisk@home inbound call problem to SIP trunk. (voipfone UK)
Hi all,
I have recently installed Asterisk@home and outbound calling is
working great. However I am strugglings with inbound calls. I have
setup a trunk for my provider, voipfone and in the inbound area on AMP
I have the following :-
user context name = 3011XXXX
context=from-pstn
dtmfmode=rfc2283
fromdomain=voipfone.co.uk
host=voipfone.co.uk
insecure=very
secret=XXXXXX
type=user
user=3011XXXX