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Yes, I think the dial does get executed (sonny calling outbound 202-555-1212): core set verbose 3 Console verbose was OFF and is now 3. -- Executing [912025551212 at from-internal:1] Log("PJSIP/sonny-00000031", "NOTICE, Dialing out from "" <sonny> to 12025551212 through fromgw") in new stack [Mar 15 19:2...

That was the issue, thanks. I now am able to get the caller ringing on an outbound call, but an external phone number (E164) I am dialing does not ring. On Sun, Mar 15, 2015 at 12:19 PM, George Joseph <george.joseph at fairview5.com > wrote: > > > On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic >> configuration works, and I am connected to a SIP trunk using SIP.US, and >> have set up my inbound calling which works correctly (when...

On Sun, Mar 15, 2015 at 1:33 PM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > Yes, I think the dial does get executed (sonny calling outbound > 202-555-1212): > > core set verbose 3 > Console verbose was OFF and is now 3. > -- Executing [912025551212 at from-internal:1] > Log("PJSIP/s...

...ing, and I am trying to configure a SIP trunk for outbound and inbound calling, and a DID for the Asterisk server, which is used for incoming calls from PSTN. I configured my SIP.US trunks (showing one gateway, gw1, here for brevity, have two: gw1 & gw2, which are both configured on my end): [sonnyGW1] type=registration transport=transport-udp outbound_auth=sonnyGW1_auth server_uri=sip:gw1.sip.us client_uri=sip:sonny at gw1.sip.us contact_user=sonny retry_interval=60 forbidden_retry_interval=600 expiration=3600 [sonnyGW1_auth] type=auth auth_type=userpass password=somepassword username=sonny...

On Sun, Mar 15, 2015 at 5:56 PM, Sonny Rajagopalan <sonny.rajagopalan at gmail.com> wrote: > George, > > I have the detailed log below. (Resending after trimming the email to 40KB.) > > The sequence below just repeats ad-nauseam. Is this a SIP trunk issue? > > Thanks! > I don't see anything obvious. T...

...network). The issue is that I am not able to make outbound calls, because the call fails with the error: res_pjsip_outbound_authenticator_digest.c:125 digest_create_request_with_auth: Unable to create request with auth.No auth credentials for any realms in challenge. CLI> pjsip show endpoint sonnyGW1 ... ========================================================================================= Endpoint: sonnyGW1 Not in use 0 of inf OutAuth: sonnyGW1_auth/sonny Aor: sonnyGW1 0 Contact: sonny...

...9:44 PM, Thyda ENG <engthyda at gmail.com> wrote: > According to what I have done , I add the message_context to the > pjsip.endpoint_custom.conf in /etc/asterisk and then I create that > message_context in the extension.conf, and it works. > > On Tue, Nov 17, 2015 at 9:34 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> So, the only thing that is needed in the endpoint definition in >> pjsip.conf (there is no such file pjsip.endpoint_custom.conf) is >> >> *message_context=astsms* >> >> Is that correct? Anyth...

Hi, I am using Asterisk 13.6.0 and was wondering if I can programmatically add users (to pjsip.conf) and dialplan (to extensions.conf) to the Asterisk server using API of some sort. Please do let me know. Thanks, Sonny. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160128/85404f6c/attachment.html>

...733 > > use_avpf=no > > ice_support=no > > media_use_received_transport=no > > trust_id_inbound=yes > > media_encryption=no > > rtp_symmetric=yes > > rewrite_contact=yes > > *message_context=astsms* > > > > On Tue, Nov 17, 2015 at 8:35 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> Hello, >> >> I am looking for documentation support for enabling instant messaging >> between endpoints using Asterisk 13.1.0 and vanilla VoIP clients such as >> Zoiper. Where do I enable this supp...

On Wed, Feb 17, 2016 at 8:56 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > I made some progress. The first thing I have realized is that it is my > Twilio configuration in pjsip_wizard.conf that was killing me. I have since > removed that entire file from /etc/asterisk and I am able to make > "...

...useam. Is this a SIP trunk issue? Thanks! --------------------- Transmitting SIP request (885 bytes) to UDP:65.254.44.194:5060 ---> INVITE sip:12025551212 at 65.254.44.194:5060 SIP/2.0 Via: SIP/2.0/UDP 18.18.19.123:5060 ;rport;branch=z9hG4bKPj70617ad5-d57a-4e5b-8043-086b0b8ebba2 From: <sip:sonny at 192.168.13.121>;tag=6953923b-1798-41a0-a983-fe27ff6a1db8 To: <sip:12025551212 at 65.254.44.194> Contact: <sip:dea9e47d-3a06-4e6e-b88f-8bac70fb6e0b at 18.18.19.123:5060> Call-ID: 012135e9-b05e-4ffd-8ed7-32b3160273e3 CSeq: 6753 INVITE Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVI...

Yes, it is enabled on port 5060. I do receive a TCP ACK back from the server, so I know the TCP segment is received at the server hosting the Asterisk build. On Wed, Feb 17, 2016 at 8:36 AM, A J Stiles <asterisk_list at earthshod.co.uk> wrote: > On Wednesday 17 Feb 2016, Sonny Rajagopalan wrote: > > OK. Let me ask this. Is anything else necessary, except choosing TCP as > the > > preferred protocol on the client, to make TCP w Asterisk work? At the > > moment, I have only changed one line in pjsip.conf from my working UDP > > setup: > > &...

On Sun, Mar 15, 2015 at 10:34 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > That was the issue, thanks. I now am able to get the caller ringing on an > outbound call, but an external phone number (E164) I am dialing does not > ring. > Any error messages? If you set 'core set verbose 3' and...

...ss: posixAccount cn: test1$ sn: test1$ uid: test1$ uidNumber: 1041 gidNumber: 515 homeDirectory: /dev/null loginShell: /bin/false description: Computer gecos: Computer Needless to the computer is not able to join the domain... Whereas a working entry migrated from tdbsam looks like this: dn: uid=sonny$,ou=computers,dc=redcircle uid: sonny$ sambaSID: S-1-5-21-1595696850-3378076689-3030227139-3008 sambaPrimaryGroupSID: S-1-5-21-1595696850-3378076689-3030227139-1201 objectClass: sambaSamAccount objectClass: account displayName: SONNY$ sambaPwdMustChange: 2147483647 sambaAcctFlags: [W ] sam...

...ch as Zoiper. Where do I enable this support on the server side and does it need anything on the client side? I see plenty of online help for chan_sip, but nothing for chan_pjsip. I imagine there is both pjsip.conf configuration and extensions.conf configuration? Any help is appreciated. Thanks, Sonny. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20151116/5667dcb0/attachment.html>

On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > I have setup my Asterisk 13.1.0 server with PJSIP on AWS/EC2. My basic > configuration works, and I am connected to a SIP trunk using SIP.US, and > have set up my inbound calling which works correctly (when I call my PBX > DI...

...ples, long ago, I removed pjsip.conf-based Twilio config and placed it all in pjsip_wizard.conf. Thanks, re: wiki, I will be using it heavily, for sure ;-) On Thu, Feb 18, 2016 at 9:56 PM, George Joseph <george.joseph at fairview5.com> wrote: > > > On Thu, Feb 18, 2016 at 7:25 PM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> Hello, >> >> I have an Asterisk 13.6.0 PBX using PJSIP connected to the Twilio >> gateway. I am able to make calls outbound through the gateway, but I am not >> able to make calls into the PBX from...

...be able to > send a call to an extension where it is behind NAT, Asterisk must update > the contact to have the current IP and port that the phone registered via > (i.e. the WAN IP of the NAT, and the WAN port that it is retaining state > for). > > On Thu, Jan 8, 2015 at 10:15 AM, Sonny Rajagopalan < > sonny.rajagopalan at gmail.com> wrote: > >> I am following the instructions in >> https://wiki.asterisk.org/wiki/display/AST/Basic+PBX+Functionality and I >> am trying to make a call from extension Alice (6001) to extension for Bob >> (6002). When...

...pt choosing TCP as the preferred protocol on the client, to make TCP w Asterisk work? At the moment, I have only changed one line in pjsip.conf from my working UDP setup: [transport-tcp] type=transport protocol=tcp ; <--------------- only this line was changed. On Wed, Feb 17, 2016 at 8:28 AM, Sonny Rajagopalan < sonny.rajagopalan at gmail.com> wrote: > OK. I will report with my findings. It appears increasingly likely that I > have done something very silly on my side. It is a little perplexing that > the EXACT setup (on the same machine) worked for UDP ... > > On Wed, F...

...t/allow = ulaw aor/qualify_frequency = 15 And--of course, I do have the DID configured on my extension, and in the dialplan "from-external" (confirmed using dialplan show from-external). What is incorrect, and what should I be doing? Any help is appreciated deeply. Thank you, Cheers, Sonny. -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160218/332b1144/attachment.html>