search for: myprovider

Displaying 20 results from an estimated 40 matches for "myprovider".

2007 Aug 03
0
Several doubts on Asterisk as an UAC
...lient* (I believe this is called UAC [User Agent Client] in the SIP world, right?) connecting to a SIP provider (such as FWD). I'm using asterisk 1.4.6, so I would like to talk *only* about configuration on asterisk 1.4.x. * SIP channels on outgoing calls If on my sip.conf I have a section [myprovider], it always creates a new SIP channel "SIP/myprovider", right? If I want to use it on extensions.conf to call extension 464646 there, I can use: Dial(SIP/myprovider/464646) or: Dial(SIP/464646 at myprovider) Is that right? If I don't want to keep the section on sip.conf, and mypro...
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/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6::user762 at myprovider.biz/${EXTEN}) exten => _XX.,n,Dial(SIP/user762:passwdk5j6 at myprovider.biz/${EXTEN}) The From part of the SIP INVITE always has the...
2016 Sep 17
2
Ast 13.11.2 : bridgepeer variable empty ?
Hello a call goes out and is answered : [Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b is making progress passing it to SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b answered SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel SIP/myprovider-0000010b joined 'simple_brid...
2004 Jan 09
1
At last!!! :)
...ine, I got rid of that "icmp udp unreachable" error. My next problem was the call stays on on Cisco gateway, but the ATA drops it. I figured out it was my mistake on dialplan in extensions.conf --- (it took me a day to notice it.. damn!). my config was: exten=>_.,1,Dial(SIP/${EXTEN}@myprovider,tr). The reason why my ATA is getting fast busy (or dropping the call immediately) while Cisco gateway (myprovider) is trying to connect my call, was that I am missing the "seconds" parameter. When I changed this to Dial(SIP/${EXTEN}@myprovider,20,tr), I was able to connect. There is...
2011 Apr 24
1
Realtime and priority labels
In the following example exten => _1NXXNXXXXXX,1,Set(GROUP(outbound)=myprovider) exten => _1NXXNXXXXXX,n,Set(COUNT=${GROUP_COUNT(myprovider at outbound)}) exten => _1NXXNXXXXXX,n,NoOp(There are ${COUNT} calls for myprovider) exten => _1NXXNXXXXXX,n,GotoIf($[ ${COUNT} > 2 ]?denied : continue) exten => _1NXXNXXXXXX,n(denied),NoOp(There are too many calls up) exten...
2020 Jan 16
1
Asterisk16 - PJSIP - Error 401 on outbound registration
...dualing pcaps at the endpoints ? No.tcpdump -nqt -s 0 -i enp0s31f6 -A "dst xxx.yyy.78.36 and dst port 5060" where xxx.yyy.78.36 is the provider Kamailio IP Capture being: IP zzz.xyz.174.138.58738 > xxx.yyy.78.36.5060: UDP, length 570 E..V.T at .?...X....2N$.r...B..REGISTER sip:sip.myprovider.net SIP/2.0 Via: SIP/2.0/UDP zzz.xyz.174.138:5060;rport;branch=z9hG4bKPj673a37a2-da52-4f8f-b460-17a93005bc98 From: <sip:123456 at sip.myprovider.net>;tag=d0be9b76-6363-4ce8-b747-7d75f222eef7 To: <sip:123456 at sip.myprovider.net> Call-ID: e906c156-a23f-4cff-b099-43c61a4447c5 CSeq: 479...
2010 Jun 02
6
How do you hangup a call without terminating your session?
...ing **. 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: Dial("SIP/14165551212@<MyProvider>,30,TgH,"") This Dial command is buried in a context that is called using the Gosub command. When I press ** the Dial command exits and processing continues as expected given the 'g' option, but when processing returns to the calling context after the Return statement is reac...
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:
2020 Jan 19
2
Asterisk16 - PJSIP - Error 401 on outbound registration
...d thus > expected to be the same? It become stranger and stranger: on one of the register peer we receive in asterisk: *CLI> [2020-01-19 15:23:18] WARNING[17469]: res_pjsip_outbound_registration.c:1021 handle_registration_response: Fatal response '401' received from 'sip:<myprovider>' on registration attempt to 'sip:<myuser>@<myprovider>', stopping outbound registration On the other one: [2020-01-19 15:23:46] WARNING[17469]: res_pjsip_outbound_registration.c:801 schedule_retry: No response received from 'sip:<other provider>' on re...
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 [provid...
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]...
2004 Nov 26
1
direct asterisk to asterisk SIP calls without external SIP provider
...ones 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 register => <telephone number>:<password>@sip.myprovider.com in sip.conf. I also included a section [mysipprovider] type=peer context=fromINTERNET host=sip.myprovider.com and used Dial(SIP/<telephone number>@mysipprovider) in my dialplan. Context fromINTERNET only consisted of exten => s,1,Dial(FXSport/0,,tH) This setup was working great, but...
2006 Feb 12
0
[ANNOUNCE] PKCS#11 support in OpenSSH 4.3p2 (version 0.07)
...at uses KDE and .NET in order to display these dialogs. You can view full usage by: $ ssh-agent /bin/sh $ ssh-add -h A common scenario is the following: $ ssh-agent /bin/sh $ ssh-add --pkcs11-ask-pin `which openssh-kde-dialogs.sh` $ ssh-add --pkcs11-add-provider --pkcs11-provider /usr/lib/pkcs11/MyProvider.so $ ssh-add --pkcs11-add-id --pkcs11-slot-type label --pkcs11-slot "MyToken" --pkcs11-id-type subject --pkcs11-id "/C=XX/CN=YY" $ ssh myhost In order to see available objects, you can use: $ ssh-add --pkcs11-show-slots --pkcs11-provider /usr/lib/pkcs11/MyProvider.so $ ssh-add...
2016 Sep 19
2
Ast 13.11.2 : bridgepeer variable empty ?
...gmludo.eu/ > > 2016-09-17 11:47 GMT+02:00 Jonas Kellens <jonas.kellens at telenet.be > <mailto:jonas.kellens at telenet.be>>: > > 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/m...
2013 Apr 10
4
ACD problem
...I want that 3rd call to be queued. I don't think that the config below will direct calls to extension 1001 because the second line states that any incomming calls should be routed to extension 1000. How do I change this so that calls are directed to?all of my?exensions? extensions.conf [from-myprovider] exten => *DID number*,1,Answer exten => *DID number*,2,Dial(SIP/1000) exten => *DID number*,3,Queue(support) ;not sure if this line belongs here exten => *DID number*,4,Hangup ? queues.conf ? [general] [support] musicclass=default strategy=rrmemory joinempty=no leavewhenempty=yes ring...
2017 Feb 06
3
Call List Campaign to an IVR
...them. That way, they were in control and things were done on > their terms. > On 6/02/2017, at 11:34 AM, Steve Edwards <asterisk.org at sedwards.com> > wrote: > > Love the idea. How? On Mon, 6 Feb 2017, Matt Riddell wrote: > exten => _X.,1,Dial(SIP/0111${EXTEN}@myprovider&SIP/1${EXTEN}@myprovider,3) Amazing. Who knew? So how/why does this work? I see 2 calls going out to my cell. Does the first 'busy out' my number at my cell provider so the second goes straight to VM? What part does the '0111' play? -- Thanks in advance, -----------------...
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
2020 Jan 20
0
Asterisk16 - PJSIP - Error 401 on outbound registration
...lt;snip> > It become stranger and stranger: on one of the register peer we receive in > asterisk: > > *CLI> [2020-01-19 15:23:18] WARNING[17469]: > res_pjsip_outbound_registration.c:1021 handle_registration_response: Fatal > response '401' received from 'sip:<myprovider>' on registration attempt to > 'sip:<myuser>@<myprovider>', stopping outbound registration > What is the actual full configuration for this? > On the other one: > > [2020-01-19 15:23:46] WARNING[17469]: > res_pjsip_outbound_registration.c:801 schedule_r...
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(${myVar}) ; <====== myVar is empty I thought it might be related to write space since call files didn't see...
2007 Nov 01
1
Call Failed
After so many rings when the party does not answer, my SIP phone says Call Failed. Why doesn't it just keep ringing? Here's the dial plan rule: exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sip.myprovider.com,,r) exten => _NXXXXXXXXX,n,Hangup()