Re: [asterisk-users] RTP handling
On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere j...@jeff.net wrote: Hello, I am wondering if asterisk does anything at all to RTP packets passed from channel to channel if no transcoding is involved? Can I assume that the packet that left phone A, arrived at the asterisk server, was copied to phone B's channel and eventually arrived at phone B had exactly (byte for byte) the same payload? Assume two SIP endpoints, no NAT involved. That will only happen when the call is natively bridged: Non-native bridge: Packets can get translated or Asterisk has an interest in the packet for things like DTMF or call recording. Native bridge doing packet-to-packet (Local bridging): Packets come in on one channel and go out the other channel with nothing else done to them. Native bridge doing direct media (Remote bridging): Packets go directly between endpoints so Asterisk never sees them. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP handling
On Tue, Mar 24, 2015 at 4:59 PM, Jeff LaCoursiere j...@jeff.net wrote: On 03/24/2015 04:28 PM, Richard Mudgett wrote: On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere j...@jeff.net wrote: Hello, I am wondering if asterisk does anything at all to RTP packets passed from channel to channel if no transcoding is involved? Can I assume that the packet that left phone A, arrived at the asterisk server, was copied to phone B's channel and eventually arrived at phone B had exactly (byte for byte) the same payload? Assume two SIP endpoints, no NAT involved. That will only happen when the call is natively bridged: Non-native bridge: Packets can get translated or Asterisk has an interest in the packet for things like DTMF or call recording. Native bridge doing packet-to-packet (Local bridging): Packets come in on one channel and go out the other channel with nothing else done to them. Native bridge doing direct media (Remote bridging): Packets go directly between endpoints so Asterisk never sees them. Richard Thanks for the quick reply RIchard! Can I force native bridging, or does it default to that if I don't configure direct media? The dialplan will be very simple - extensions calling extensions within a context. No DTMF, no recording, no mixing for conference, etc. You cannot force native bridging. It will switch to native bridging if you don't set anything that makes Asterisk interested in the media stream. Such as enabling DTMF features in features.conf and Dial flags like t or T. Richard -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] RTP handling
On 03/24/2015 04:28 PM, Richard Mudgett wrote: On Tue, Mar 24, 2015 at 4:17 PM, Jeff LaCoursiere j...@jeff.net mailto:j...@jeff.net wrote: Hello, I am wondering if asterisk does anything at all to RTP packets passed from channel to channel if no transcoding is involved? Can I assume that the packet that left phone A, arrived at the asterisk server, was copied to phone B's channel and eventually arrived at phone B had exactly (byte for byte) the same payload? Assume two SIP endpoints, no NAT involved. That will only happen when the call is natively bridged: Non-native bridge: Packets can get translated or Asterisk has an interest in the packet for things like DTMF or call recording. Native bridge doing packet-to-packet (Local bridging): Packets come in on one channel and go out the other channel with nothing else done to them. Native bridge doing direct media (Remote bridging): Packets go directly between endpoints so Asterisk never sees them. Richard Thanks for the quick reply RIchard! Can I force native bridging, or does it default to that if I don't configure direct media? The dialplan will be very simple - extensions calling extensions within a context. No DTMF, no recording, no mixing for conference, etc. Cheers, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] outbound calls
hi the issue still the same i have 2 trunks whe i configure the first in x-lite and the second in my server or my ip-phone snom320 directly from x-lite i can call my trunk without issue but when i try ti call from snom320 to x-lite or from my server asterisk using extension in x-lite the call all time is failed any help please thanks and regards 2015-03-20 19:28 GMT+00:00 Trey Hilyard kct...@gmail.com: So you are saying that it resolved the issue to activate voicemail on the device that sits past your trunk provider? That confuses me a little, but if your calls are working, that's great news. On Fri, Mar 20, 2015 at 1:44 PM Salaheddine Elharit salah.elharit...@gmail.com wrote: i noticed that when i active the voicemail in the IP-phone where the number 0033149xx is configured i can call this number without issue Using SIP RTP TOS bits 184 == Using SIP RTP CoS mark 5 -- Called SIP/FD/0033149xx == Begin MixMonitor Recording SIP/101-010d -- SIP/FD-010e is making progress passing it to SIP/101-010d 0x2b393cfc2610 -- Probation passed - setting RTP source address to 192. 168.1.138:55542 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 -- SIP/FD-010e answered SIP/101-010d 0x1d08efa0 -- Probation passed - setting RTP source address to 217.195.xx.xx:46346 thanks and regards. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 13.1.0/PJSIP outbound calling using SIP trunk: Unable to create request with auth.No auth credentials for any realms in challenge.
Hi George, Well, as it turns out the removal of realm in sonnyGW1_auth above does not remove the issue. I still see the issue. I did not see the issue earlier likely due to the CLI logging command mixup which I have now solved using a wireshark trace (CLI was just too verbose). I see the 407 authentication required still, and the following pattern just repeats at the Asterisk server (which is connected to the SIP trunk at 65.254.44.194) because the SIP trunk needs it to complete the outbound call, but the Asterisk server doesn't ever send it even after the 407 from the SIP trunk: Wireshark trace of failed outbound call: 217274 5915.986472000 sonnysMachine 65.254.44.194 SIP/SDP 1227 Request: INVITE sip:16175551212@65.254.44.194:5060 | 217280 5916.059148000 65.254.44.194 sonnysMachine SIP 385Status: 100 Trying | 217282 5916.059909000 65.254.44.194 sonnysMachine SIP 582Status: 407 Proxy Authentication Required | 217285 5916.060227000 sonnysMachine 65.254.44.194 SIP 425Request: ACK sip:16175551212@65.254.44.194:5060 | ... (repeats ad infinitum) When I look at the challenge in 407 Proxy Authentication Required from the server, I see that the realm is 65.254.44.194 (gw1.sip.us), but the appropriate Authorization (sent in the trunk registration, for example) is never sent back from the Asterisk server. Here's what the SIP trunk actually says (407 Auth required message; the nonce was changed by me): Wireshark detail of 407 Proxy Authentication Required packet from SIP trunk: Proxy-Authenticate: Digest realm=65.254.44.194, nonce=BLUBBERbbb/e495019b-83b4-491c-8f33-3e238a3c6af2, qop=auth Authentication Scheme: Digest Realm: 65.254.44.194 Nonce Value: BLUBBERbbb/e495019b-83b4-491c-8f33-3e238a3c6af2 QOP: auth And here's how the SIP trunk registration works (correctly); note the bigger REGISTER message in the 3rd line pertaining to the registration at 65.254.44.194, it pertains to the additional 274 bytes of authentication information: Wireshark detail of successful SIP trunk registration: 12634 230.39042 sonnysMachine 65.254.44.194 SIP 543Request: REGISTER sip:gw1.sip.us(fetch bindings) | 12635 230.461572000 65.254.44.194 sonnysMachine SIP 560Status: 401 Unauthorized(0 bindings) | 12637 230.462041000 sonnysMachine 65.254.44.194 SIP 815Request: REGISTER sip:gw1.sip.us(fetch bindings) | 12639 230.53510 65.254.44.194 sonnysMachine SIP 486Status: 200 OK(0 bindings) | Any help is deeply appreciated. Has anyone successfully done SIP trunk registration with PJSIP in Asterisk 13.1.0? On Sun, Mar 15, 2015 at 12:34 PM, Sonny Rajagopalan sonny.rajagopa...@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. On Sun, Mar 15, 2015 at 12:19 PM, George Joseph george.jos...@fairview5.com wrote: On Sun, Mar 15, 2015 at 8:32 AM, Sonny Rajagopalan sonny.rajagopa...@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 DID, the call does come into my PBX 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: sonnyGW1Not in use 0 of inf OutAuth: sonnyGW1_auth/sonny Aor: sonnyGW1 0 Contact: sonnyGW1/sip:65.254.44.194:5060 Unknown nan Transport: transport-udp udp 0 0 0.0.0.0:5060 Identify: sonnyGW1/sonnyGW1 Match: 65.254.44.194/32 My pjsip.conf is as below [sonnyGW1] type=registration transport=transport-udp outbound_auth=sonnyGW1_auth server_uri=sip:gw1.sip.us client_uri=sip:so...@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 realm=gw1.sip.us You probably need to remove the 'realm' line so that it will match any realm in the challenge. [sonnyGW1] type=aor contact=sip:65.254.44.194:5060 [sonnyGW1] type=endpoint transport=transport-udp context=gateway1 allow=!all,ulaw outbound_auth=sonnyGW1_auth aors=sonnyGW1 [sonnyGW1] type=identify endpoint=sonnyGW1 match=65.254.44.194 My
[asterisk-users] RTP handling
Hello, I am wondering if asterisk does anything at all to RTP packets passed from channel to channel if no transcoding is involved? Can I assume that the packet that left phone A, arrived at the asterisk server, was copied to phone B's channel and eventually arrived at phone B had exactly (byte for byte) the same payload? Assume two SIP endpoints, no NAT involved. Thanks, j -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
