search for: siptrunk

...p i have "match_auth_username=yes" and have nothing in pjsip. I have a lot of endpoints and registrations on same SIP server. And it's problem in pjsip now. Is not it? I requesting to add new value for endpoint option identify_by. The value 'uri'. Simple config (cutted): [siptrunk] type=registration transport=udp-transport outbound_auth=siptrunk server_uri=sip:sip.example.com client_uri=sip:1234567890 at sip.example.com retry_interval=60 contact_user=siptrunk-in [siptrunk-in] type=endpoint transport=udp-transport context=from-trunk disallow=all allow=ulaw outbound_auth=sipt...

I have the following extension to dial outside via SIP it's like this: phone----asterisk-----internet-----SIP provider----USA exten => _91NXXNXXXXXX,1,AGI(call_log.agi,${EXTEN}) exten => _91NXXNXXXXXX,2,Dial(${SIPtrunk}/${EXTEN},55,o) exten => _91NXXNXXXXXX,3,Hangup I want to strip the digit 9 before sending it to the SIP provider. Also, any suggestions for the above definition? Thanks, Erick. --

...pjsip. > > I have a lot of endpoints and registrations on same SIP server. > And it's problem in pjsip now. Is not it? > > I requesting to add new value for endpoint option identify_by. The > value 'uri'. > Simple config (cutted): > > [siptrunk] > type=registration > transport=udp-transport > outbound_auth=siptrunk > server_uri=sip:sip.example.com <http://sip.example.com> > client_uri=sip:1234567890 at sip.example.com <mailto:client_uri=sip:1234567890 at sip.example.com> > retry_inte...

Hi list, I have to connect two asterisk box, in this scenario: [asterisk1]----sip----[asterisk2]----PSTN I must use sip, cos we'll use cisco rtp header-compression to save bandwidth. Could you tell me the best way to send calls from asterisk1 to asterisk2, since I cannot use IAX trunking? Thanks in advance Eduardo

...) exten => s,3,WaitExten() The ringing sound doesn't work for any extension if I use this one. I just get silence until someone answers. How come? I use Asterisk 1.4.10. I have attached my extensions.conf file to this email. Thanks, Simon -------------- next part -------------- [globals] SIPTRUNK=4185555555 IAXTRUNK=5145555555 [default] exten => s,1,Answer() exten => s,2,Background(viagenie) exten => s,3,WaitExten() exten => i,1,Background(invalid) exten => i,n,Goto(s,1) exten => t,1,Background(please-try-again) exten => t,n,Goto(s,1) [phones] exten => 101,1,Dial...

...uot; and have nothing in pjsip. > > I have a lot of endpoints and registrations on same SIP server. And it's > problem in pjsip now. Is not it? > > I requesting to add new value for endpoint option identify_by. The value > 'uri'. > Simple config (cutted): > > [siptrunk] > type=registration > transport=udp-transport > outbound_auth=siptrunk > server_uri=sip:sip.example.comclient_uri=sip:1234567890 at sip.example.com > retry_interval=60 > contact_user=siptrunk-in > > [siptrunk-in] > type=endpoint > transport=udp-transport > context=...

Thanks George, for your mighty quick response. I made the changes (re: server_uri_pattern etc.) and still, no luck--it fails for the same error. BTW, there is nothing for transport (but this is the same config from my SIP/UDP + Twilio days, which worked): *CLI> pjsip show transport twilio-siptrunk Unable to find object twilio-siptrunk. *CLI> pjsip show identifies No objects found. I did add ;transport=tcp to my Origination URI after wireshark revealed everything was received as UDP into Asterisk, so we can rule out that issue (I confirmed that I am getting TCP based SIP INVITEs from Twi...

I have a puzzling situation, and would be grateful for any insight. I have a dialplan that forwards an incoming call out to another number via the same SIP trunk as it came in on. e.g. [from-siptrunk] exten => 0123456789,1,NoOp exten => 0123456789,n,Dial(SIP/siptrunk/0987654321) Now, if I use a different SIP trunk for the outbound call, than the inbound call came on, the call is set up fine - the Answer signal from the called party gets propagated back to the caller, and they can hear ea...

I have 10 SIP trunks that I'd really like to round-robin load balance. Currently I have a macro that switches between available lines, but there really must be a function in Asterisk to do this on its own. So my question is just that, are there any easy ways for Asterisk to either balance between SIP trunks or even just a built in function to find the next available SIP trunk. I think using

...> >> I have a lot of endpoints and registrations on same SIP server. And it's >> problem in pjsip now. Is not it? >> >> I requesting to add new value for endpoint option identify_by. The value >> 'uri'. >> Simple config (cutted): >> >> [siptrunk] >> type=registration >> transport=udp-transport >> outbound_auth=siptrunk >> server_uri=sip:sip.example.comclient_uri=sip:1234567890 at sip.example.com >> retry_interval=60 >> contact_user=siptrunk-in >> >> [siptrunk-in] >> type=endpoint >&g...

Hi all. Is it possible to get * to send calls to different sip trunks depending on what codec the incoming call use? This to avoid transcoding Anders -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050924/0e209878/attachment.htm

...gress already, and it didn't help. But thanks for the suggestion! Tony > I have a puzzling situation, and would be grateful for any insight. > > I have a dialplan that forwards an incoming call out to another > number via the same SIP trunk as it came in on. e.g. > > [from-siptrunk] > exten => 0123456789,1,NoOp > exten => 0123456789,n,Dial(SIP/siptrunk/0987654321) > > Now, if I use a different SIP trunk for the outbound call, than the > inbound call came on, the call is set up fine - the Answer signal from the > called party gets propagated back to th...

Hi! I'm trying to set up a SIP trunk so that I can test calls, etc., between a new Asterisk box, and an old 1.4 box. --------------------------------------------------------------------------- New box: root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf siptrunk.conf: [box1] ; All box1 extensions; see extensions.conf type=peer context=adhearsion host=172.17.0.17 ; IP for old system disallow=all allow=g729 canreinvite=yes qualify=no Old box: root at asterisk1:/etc/asterisk# head -1 sip.conf #include siptrunk.conf siptrunk.conf: [box2] ; A...

...ly the negative. An asterisk indicates a 404 error. On Tue, Mar 5, 2019 at 12:51 PM Doug Lytle <support at drdos.info> wrote: > > On 3/5/19 2:46 AM, Gokan Atmaca wrote: > > Asterisk can send calls, but I don't get a call. What could be the problem? > > > > [from-siptrunk] > > exten => 13XXXXXXX,1,dial(${OPERATOR},20) > > > > exten => _13XXXXXXX,1,dial(${OPERATOR},20) > > Doug > > Doug > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://ww...

...ying on since I have other calls coming in on the same T1. If I spy on a SIP extension instead of a SIP trunk, I hear both sides just fine. I am using a recent version of Asterisk 1.2 and I am using g729 licenses. This is the command I am using to spy. exten => 8011,1,ChanSpy(Sip/SIPTRUNK|bqv(4)) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070926/7854ef8f/attachment.htm -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type:...

Hello Asterisk can send calls, but I don't get a call. What could be the problem? [from-siptrunk] exten => 13XXXXXXX,1,dial(${OPERATOR},20) Thanks.

...m using the new SipAddHeader application on Asterisk 1.2.1, here's a snip of my extensions: exten => _9XXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM} exten => _9XXXXXXX,2,SipAddHeader(P-Asserted-Identity: tel:${CALLERIDNUM}) exten => _9XXXXXXX,3,Dial(SIP/${EXTEN}@${SIPTRUNK},,tT) exten => _9XXXXXXX,4,Congestion The problems is that Asterisk only add the last (priority 2) header if I remove the second SipAddHeader the first one appears on the INVITE. How do I add two or more headers ? TIA -- Juanjo sin .sig :(

Am 22.11.2014 um 12:51 schrieb Andreas Sikkema: >> but as soon as I configure another sip registration on another server, >> outgoing >> calls drop after 32 seconds. > Are both your servers behind the same NAT router? > thanks for taking part... I don?t know... one is siptrunk.ovh.net and the other one is sip.ovh.fr how can i determine and how could that affect... I mean... why do they interfere at all? thanks, yves --- Diese E-Mail wurde von Avast Antivirus-Software auf Viren gepr?ft. http://www.avast.com

...gt; >> but as soon as I configure another sip registration on another server, > >> outgoing > >> calls drop after 32 seconds. > > Are both your servers behind the same NAT router? > > > thanks for taking part... > > I don?t know... > one is > > siptrunk.ovh.net > > and the other one is > > sip.ovh.fr > > how can i determine and how could that affect... I mean... why do they > interfere at all? > > thanks, > yves > > --- > Diese E-Mail wurde von Avast Antivirus-Software auf Viren gepr?ft. > http://www.avast...

Hi! Is there a simple way for an * newbie to force * to use different sip-trunks for different calls. I have 2 siptrunks, one for inland calls and one for international calls. All in country numbers starts with 0 and all international starts with 00. This I have configured in the outbound routing. But * always use the incountry trunk because the 0. dialpattern is also true for international calls How to fix t...