Displaying 20 results from an estimated 1000 matches similar to: "Problems with outbound calls through VSP"
2007 May 14
1
Difference between making a call and Originate
When I make a regular call from my SIP phone connected to my Asterisk
server I have no issues, however when I make a call using Originate :
'Channel'=>"SIP/1XXXXXXXXXX@sip.broadvoice.com",
'Context'=>'mycontext',
'Exten'=>'899',
'Priority'=>1,
'Callerid'=>'whatever'));
It creates a screech sound when the
2007 Jan 24
3
setting up AMD
I'm trying get this working. I've looked through the list, and can't see
how to get AMD to print out more. I have it call and say Hello like I
normally would. I've tried to say more and less doesn't seem to matter.
After I hangup it does recognize hangup. Here's logging during an attempt
where I make outbound call and answer, but then hangup after 1-2 seconds:
Jan 24
2014 Jan 31
2
callfiles.call
hello list,
i have created a callfiles with my asterisk 1.4.43 like:
Channel: SIP/watara/06xxxxxxxx
MaxRetries: 10
RetryTime: 5
WaitTime: 20
Context: mycontext
Extension: s
Priority: 1
extensions.conf
mycontext
exten => s,1,Ringing()
exten => s,n,Playback(hello-world)
exten => s,n,Dial(SIP/105)
exten => s,n,Hangup()
it works with one number how can i do in order to create a
2004 Jul 22
4
VSP? Looking for advice.
Has anyone tried using BroadVoice for VSP? I have Asterisk configured
for a home office & I've been trying to decide which VoIP provider to go
with for a little while now. I had heard you could get sub $.01 calls
but I have not found that to be true yet (not saying it's not possible,
I just haven't found it!).
Also I'm not sure if BV will support multiple lines. Any
2004 Jul 17
1
voicemail broadcast feature
Using CVS from 7/12/04 and trying to get the voicemail broadcast feature
to work.
Voicemail.conf has
[mycontext]
3722 => 1234,BroadCast Test,,,cc=*@mycontext
.
then many other voicemail boxes.
-----
whenever I leave voicemail at box 3722, only box 3722 gets the
voicemail. It is not expanding it to other voicemail boxes in the
[mycontext] context.
Even if I replace the cc= line with
2005 May 26
2
voicemail comprehension
Hi all,
In order to do loadbalancing between my two *, i wanted to stock all
things concerning voicemail on a NFS partition...
I see that the voicemail system put his files onto two differents
directories :
/var/spool/asterisk/voicemail/mycontext etc.
and
/var/lib/asterisk/voicemail/mycontext etc.
I've two questions :
Why ?
and how can i do to centralize the destination of the messages AND
2003 Apr 28
1
Turning off Bridging?
Hi folks
Is it possible to turn off the native bridging on Asterisk?
I've been hacking about app_disa.c to support account & pin numbers, that tag the calls
depending on who logs in.....
It all works fine, then dials the destination number they requested.
My setup is as follows
[ENDPOINT] <IAX1> [MYASTERISKBOX] < IAX1 > [TELCOBOX]<>(PSTN)
If i dial
2017 Nov 22
3
Chan Local, Originate and slin
Hi all!
Asterisk 13.1.0 Ubuntu 16.04, all latest.
Can anybody explain this to me - I run Originate command from dialplan:
same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum})
and I get crazy sound distortion in the conference, and I see that
transcoding takes place here:
NativeFormats: (slin192)
WriteFormat: slin
ReadFormat: slin192
WriteTranscode: Yes (slin
2009 Dec 21
1
Incoming calls coming into default context
My SIP-provider sends my a SIP-invite like this :
INVITE sip:329298yyy6 at 80.XX.XX.69:5060 SIP/2.0
Via: SIP/2.0/UDP 80.XX.XX.68:5060;branch=z9hG4bKf395877e02e5aa21fd8f5a0c
Max-Forwards: 70
From: <sip:321445xxx6 at 80.XX.XX.69>;tag=f395877e02bf8eb2fd8f5a0e
To: <sip:329298yyy6 at 80.XX.XX.69>
Call-ID: f395877e02187250fd8f5a0f at 80.XX.XX.68
CSeq: 1 INVITE
User-Agent: SysMaster VoIP
2013 Nov 05
1
[LLVMdev] Thread-safe cloning
Sorry to resurrect an old thread, but I finally got around to testing
this approach (round tripping through bitcode in memory) and it works
beautifully - and isn't that much slower than cloning.
I have noticed however that the copy process isn't thread-safe. The
problem is that in Function, there is lazy initialization code for
arguments:
void CheckLazyArguments() const {
if
2017 Dec 13
2
Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?
Currently using PJSIP. First, they want me to get this working with the existing PJSIP configuration, but then setup a second box using chan_sip performing similar work.
For PJSIP...
I currently have an endpoint configured to a system using IP based authentication. It is configured with a match setting in the endpoint section.
All channels coming from that IP address go to this endpoint.
They
2004 Oct 06
1
how does agent logoff if you supply extension?
Per the wiki:
Logging off
1. call the extension for AgentCallbackLogin
2. enter your password followed by #
3. when asked for the extension number just press #
But if your exten=> is this:
exten => 2010,1,AgentCallbackLogin(3333|3044@mycontext)
How do they logoff per the wiki's directions? If you use ACBL as above, it
never asks you for the extension number because you have
2012 Aug 05
3
Voice Mail beep / tone detection
Though asterisk support AMD which is based on silence detection but I did
not found support of tone / beep detection in asterisk to record a voice
message for answering machines after detecting tone
Will appreciate any help in this regard
Best Regards
*Tahir Almas*
Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT
Unified Communication Telemarketing
2018 Mar 14
2
PJSIP Originate
I am using AMI Originate to perform a new outbound call.
The SIP Provider we send the call to wants us to pass the caller id of the person we are calling for in the Contact header.
For the AMI Originate, I pass the caller id information data in the CallerID field. However, this is never being passed through the PJSIP INVITE header
Action: Originate
ActionID: S598
Channel: PJSIP/133 at 1002
2011 Feb 04
0
[LLVMdev] ConstantBuilder proposal
If you remove all the 'static's from the member functions, it'd work
more like IRBuilder.
It would also allow you to take the LLVMContext& as a constructor
parameter, so that methods like this:
On Fri, Feb 4, 2011 at 6:57 PM, Talin <viridia at gmail.com> wrote:
> /// GetStruct - return a constant struct given a context and a vector
> /// of elements.
> static
2007 Feb 27
1
Not registering Port with VSP
Hello All,
For some reason my asterisk server is not registering a port number with
my VSPs. This is causing problems where people are not able to dial in
from any of my SIP or IAX VSPs.
I do have one VSP that has hard coded my IP and port so I can get
incoming calls but this still leaves a problem with my other VSPs.
Hose can I get asterisk to register my IP and port? I have been
2004 Nov 30
2
* Compatible VSP Service in Ukraine?
I'm sure this might not be the correct place to ask and I have done a Google
but I can't seem to find anything that says there is a VSP that will work
with * in the Ukraine.
I have a friend that lives in Kiev and basically want a phone number there
to be able to talk to him and have him call me.
If anyone has any information on it and they are willing to share please
advise.
2007 Jan 08
0
Allowing inbound VoIP Calls from VSP
Hi All,
I think I have missed something as I am resisted with 4 VSPs and I can
not dial in using any one of them using the corresponding VoIP numbers
assigned with the VSP. I can make outbound calls to another VoIP number
to the same provider.
The weird thing is that I have a DID with a VSP and I have that working
fine but try using the associate VoIP number and nothing happens.
When
2007 Jan 12
1
Not Registering Port with VSP.
Hi All,
I seem to be having a problem with all my VSPs. When I am registering
with them I don't seem to be passing my port number. This problem
causes other users the inability to call my VoIP number with the VSP.
My VSP showed me what they are seeing.
I have changed my useragent to be: Linksys/SPA941-4.1.15
Linksys/SPA941-4.1.15 Contact sip:1234321234@aa.bb.cc.dd with no
2007 Sep 04
1
VSP authentication to incorrect context
All, I'm hoping someone can direct me as to why when someone calls my
DID Asterisk tries to authenticate the incoming call on my outbound
context. If I remove the GoTalk context I can receive incoming calls.
Outbound calls work fine while I have the GoTalk context in place.
The error I am getting when someone calls the DID is
WARNING[16072]: chan_sip.c:8272 check_auth: username mismatch,