similar to: Dialing Out through Provider with Authentication

Displaying 20 results from an estimated 300 matches similar to: "Dialing Out through Provider with Authentication"

2017 Apr 03
3
Define SIP fromuser field in Dial()-command
Hello how can I set the fromuser field of the SIP INVITE when using the Dial()-command in the dialplan ? None of the below Dial() command give the correct result : exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz) exten => _XX.,n,Dial(SIP/${EXTEN}:passwdk5j6::user762 at myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user762 at
2016 Sep 17
2
Ast 13.11.2 : bridgepeer variable empty ?
Hello a call goes out and is answered : [Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b is making progress passing it to SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b answered SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel SIP/myprovider-0000010b joined
2004 Jan 09
1
At last!!! :)
I can smile now. I made my * work with my Cisco. Finally. First problem was Ethernet on my Linux. After installing * on a different machine, I got rid of that "icmp udp unreachable" error. My next problem was the call stays on on Cisco gateway, but the ATA drops it. I figured out it was my mistake on dialplan in extensions.conf --- (it took me a day to notice it.. damn!). my config
2014 Sep 02
3
PJSIP issues with handling incoming calls
Hello guys. Have 2 external numbers that required registration on provider server, trunk1: 73432260005 at 80.75.132.66 trunk2: 73432260050 at 80.75.132.66 Thing is I can?t figure out how to route them to different IVRs by default Asterisk can?t match endpoint Request from '<sip:+ 73432260005 at 80.75.132.66>' failed for '80.75.132.66:5060' (callid:
2011 Apr 24
1
Realtime and priority labels
In the following example exten => _1NXXNXXXXXX,1,Set(GROUP(outbound)=myprovider) exten => _1NXXNXXXXXX,n,Set(COUNT=${GROUP_COUNT(myprovider at outbound)}) exten => _1NXXNXXXXXX,n,NoOp(There are ${COUNT} calls for myprovider) exten => _1NXXNXXXXXX,n,GotoIf($[ ${COUNT} > 2 ]?denied : continue) exten => _1NXXNXXXXXX,n(denied),NoOp(There are too many calls up) exten =>
2010 Jun 02
6
How do you hangup a call without terminating your session?
Asterisk 1.6 CentOS 5.0 All - I'd like to offer my users the ability to hangup a call by pressing **. I'm using an attendant, so when ** is dialled I'd like processing to return to the attendant so the user can place a subsequent call. I have setup features.conf to include: [featuremap] disconnect => ** My Dial command looks like this:
2020 Jan 19
2
Asterisk16 - PJSIP - Error 401 on outbound registration
Le 19/01/2020 à 00:31, Joshua C. Colp a écrit : > On Sat, Jan 18, 2020 at 1:14 PM Administrator <admin at tootai.net > <mailto:admin at tootai.net>> wrote: > > > Le 17/01/2020 à 11:54, Administrator a écrit : > > > > Le 15/01/2020 à 19:24, Administrator a écrit : > >> Hi all, > >> > >> we face a strange
2020 Jan 16
1
Asterisk16 - PJSIP - Error 401 on outbound registration
Le 15/01/2020 à 19:50, C.Maj a écrit : > On 2020-01-15 11:24, Administrator wrote: > > 8<'s > >> One of the provider took a pcap and told us that expiration was set to 0 >> that's why they don't accept the registration. We took a pcap on our >> side when SIP packet goes out of our server and we see that the >> expiration parameter is setted to
2007 Aug 07
1
Use of context=... in [default] section of sip.conf
Hi, If I have [myprovider] section with context=something. When I do an outgoing call by using Dial(SIP/myprovider/464646)", does context=... affect anything? As I understand it, it only affects incoming calls, but I might be wrong. Another thing, the setting of context=... on [default] section will affect all [provider] sections without context=..., right? What if I don't specify any
2015 Apr 07
1
exten versus EXTEN
p 176 has exten => 1NXXNXXXXXXX,1,Dial(SIP/${EXTEN}@myprovider) how is "exten" distinct from "EXTEN"? What is this line of code doing? https://wiki.asterisk.org/wiki/display/AST/Asterisk+Standard+Channel+Variables says that EXTEN is the current extension. In ruby, you this: H = Hash["a" => 100, "b" => 200] The => is a mapping, or at least
2013 Apr 10
4
ACD problem
? Hi, I am working on a small inbound call center solution that uses an ACD system. I might add an IVR system later on. I only have 2 extensions set up (extensions 1000 and 1001), I want the system to put new calls in a queue if both extensions are busy. I am currently subscribed with a SIP trunk provider and can successfully recieve calls. I want?to design a system where customers?can call my
2017 Feb 06
3
Call List Campaign to an IVR
> On Mon, 6 Feb 2017, Tech Support wrote: > > We were able to develop a feature to send the call to voicemail about 90% of the time. That way, an end user could (1) not be bothered by having to answer the call, (2) > delete the message without listening to it, or (3) listen to the message when it was most convenient for them. That way, they were in control and things were
2004 Nov 26
1
direct asterisk to asterisk SIP calls without external SIP provider
Hi all, I have a small system of two hardware boxes (residential gateways) running Linux with Asterisk on them. Each RG has some FXS ports to which analog telephones can be connected. I already had a working system including an external SIP provider, where both RGs would register to that provider with a telephone number and they could call each other via that telephone number. Each RG had a line
2020 Jan 15
4
Asterisk16 - PJSIP - Error 401 on outbound registration
Hi all, we face a strange behavior while connecting an Asterisk16 instance with PJSIP to 2 providers: we receive error 401 Unauthorized, both of them having Kamailio as front-end. With other providers -we don't know if they run kamailio- registration is just fine. One of the provider took a pcap and told us that expiration was set to 0 that's why they don't accept the
2017 Feb 06
4
Call List Campaign to an IVR
On Mon, 6 Feb 2017, Tech Support wrote: > We were able to develop a feature to send the call to voicemail about > 90% of the time. That way, an end user could (1) not be bothered by > having to answer the call, (2) delete the message without listening to > it, or (3) listen to the message when it was most convenient for them. > That way, they were in control and things were
2007 Jan 19
1
Set Parameter of Call Files
Hi all, I'm implementing call files and everything works nicely except that the variable that I set in the call file does not seem to get populated. Channel: SIP/MyProvider/9105555555 MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: myCallFileContext Extension: s Priority: 1 Set: myVar=MyNewValue and... [myCallFileContext] exten=>s,1,NoOp(${myVar}) ; <====== myVar is empty
2007 Nov 01
1
Call Failed
After so many rings when the party does not answer, my SIP phone says Call Failed. Why doesn't it just keep ringing? Here's the dial plan rule: exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sip.myprovider.com,,r) exten => _NXXXXXXXXX,n,Hangup()
2007 Nov 12
1
sip_chan - how to use value of the SIP 'To:' header field for extension logic
Hi, I have the following situation. I have one account created in my VoIP provider. Asterisk registers this account with the usage of 'register = ' command in the sip.conf file. I have a number of aliases assigned to my user which correspond to different public/PSTN numbers through which I am accessible. When there is an incoming call from my sip provider 'some_extension' which
2013 Apr 18
5
Dynamic realtime + queues
Hi, ? I am trying to store queues.conf to a MySQL database using dynamic realtime. I have a working ODBC connection and the queueing system already works but I want to store the queues.conf file to a database. I am following the guide from Asterisk the definitive guide, the ebook can be found at: http://ofps.oreilly.com/titles/9781449332426/asterisk-DB.html ? I have a database called asterisk
2020 Jan 18
2
Asterisk16 - PJSIP - Error 401 on outbound registration
Le 17/01/2020 à 11:54, Administrator a écrit : > > Le 15/01/2020 à 19:24, Administrator a écrit : >> Hi all, >> >> we face a strange behavior while connecting an Asterisk16 instance >> with PJSIP to 2 providers: we receive error 401 Unauthorized, both of >> them having Kamailio as front-end. With other providers -we don't >> know if they run