search for: myprovider

...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...

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...

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...

...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...

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...

...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...

...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...

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:

...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...

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...

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]...

...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...

...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...

...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...

...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...

...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, -----------------...

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

...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...

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...

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()