search for: maxptime

...ntent-Type: application/sdp Content-Length: 286 v=0 o=- 15363811 15363814 IN IP4 192.168.3.99 s=Asterisk c=IN IP4 80.142.13.32 t=0 0 m=audio 51822 RTP/AVP 8 3 112 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:112 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <--- Received SIP request (1302 bytes) from UDP:217.10.79.9:5060 ---> INVITE sip:2636146e0 at 80.142.13.32:55060 SIP/2.0 Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK56ac.c519528374e3101a61791cf5a0ad1aae.0 Via: SIP/2.0/UDP 172.20.40.6;branch=z9hG4bK56ac.2ecd3532ae51c927dabcc6e...

...body from SIP carrier) v=0 o=msw.chance4minutes.net 1234 0 IN IP4 38.126.208.46 s=sip call c=IN IP4 38.126.208.46 t=0 0 m=audio 30552 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=maxptime:20 (inside 200 OK response body from asterisk) v=0 o=root 835643920 835643920 IN IP4 201.234.196.171 s=Asterisk PBX 11.10.0 c=IN IP4 201.234.196.171 t=0 0 m=audio 12112 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv...

ping times are fine as well: [root at freepbx asterisk]# ping sipgate.de PING sipgate.de (217.10.79.9) 56(84) bytes of data. 64 bytes from sipgate.de (217.10.79.9): icmp_seq=1 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=2 ttl=57 time=46.4 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=3 ttl=57 time=46.7 ms 64 bytes from sipgate.de (217.10.79.9): icmp_seq=4 ttl=57

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

...et talking to Asterisk, which in turn puts the call through to an ITSP. The handsets likes to send audio in 40ms frames (even though Asterisk requests 20ms frames with a ptime header in the SDP). The ITSP doesn't request any particular frame length (with ptime) or state a maximum length (with maxptime), so when Asterisk receives the 40ms media frames from the handset, it simply relays it on to the ITSP. Unfortunately the ITSP doesn't support this, and the result is one-way audio. I would like to know whether there is a way to force Asterisk to repacketize the media stream, converting from 4...

...ation/sdp > Content-Length: 239 > > v=0 > o=- 1014372762 1014372762 IN IP4 192.168.13.121 > s=Asterisk > c=IN IP4 18.18.19.123 > t=0 0 > m=audio 11614 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=maxptime:150 > a=sendrecv > > [Kip-192.168.13.121*CLI> [0K<--- Received SIP response (342 bytes) from > UDP:65.254.44.194:5060 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 18.18.19.123:5060;rport=5060;branch=z9hG4bKPj70617ad5-d57a-4e5b-8043-086b0b8ebba2;received=18.18.19.123...

We have a customer with a system rejecting calls from Asterisk. It's indicating the ptime is 60, but the system admin is saying they only support 20. They are running asterisk 16.2.1 and using chan_sip Is there a way to specify this with chan_sip? Also, for my own curiosity, is there a way to specify this with PJSIP? (Trying to migrate customers to PJSIP, but we are holding until asterisk

...ttp://www.webrtc.org/experiments/rtp-hdrext/abs-send-time // a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2179369454 cname:SvzCJjIAujxHGm9P a=ssrc:2179369454 msid:Kqg5QpXyqNeviT8qxUIRi8QNUaV7mUnFiDIF add6e533-c83d-42f2-b487-fcac8646ad32 a=ssrc:2179369454 mslabel:Kqg5QpXyqNeviT8qxUIRi8QNUaV7mUnFiDIF a=ssrc:2179369454 label:add6e533-c83d-42f2-b487-fcac8646ad32 a=sendrecv a=rtcp:11077 a=rtcp-mux...

...norefersub Session-Expires: 1800 in-SE: 90 Content-Type: application/sdp Content-Length: 239 v=0 o=- 1014372762 1014372762 IN IP4 192.168.13.121 s=Asterisk c=IN IP4 18.18.19.123 t=0 0 m=audio 11614 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv [Kip-192.168.13.121*CLI> [0K<--- Received SIP response (342 bytes) from UDP: 65.254.44.194:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 18.18.19.123:5060 ;rport=5060;branch=z9hG4bKPj70617ad5-d57a-4e5b-8043-086b0b8ebba2;received=18.18.19.123 From: <sip:sonny at 192.168.1...

...STER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: multipart/mixed ;boundary=unique-boundary-1 Content-Length: 645 --unique-boundary-1 Content-Type: application/SDP v=0 o=- 95 1 IN IP4 192.168.45.194 s=- t=0 0 m=audio 5234 RTP/AVP 0 8 c=IN IP4 192.168.45.197 a=ptime:20 a=maxptime:20 a=sendrecv --unique-boundary-1 Content-Type: application/x-nt-mcdn-frag-hex ;version=sse-4.00.31 ;base=x2611 Content-Disposition: signal ;handling=optional 05006702 0107130081900000a2 09090f00e9a083000100e7 1315070011fa0f00a10d02010102020100cc046605123c --unique-boundary-1 Content-Type: appli...

...a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv The reply SDP is: v=0 o=default 1359060187 1359060187 IN IP4 10.10.22.246 s=Asterisk PBX 10.7.1 c=IN IP4 10.10.22.246 t=0 0 m=audio 32000 RTP/AVP 8 101 a=sendrecv a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=maxptime:90 Any suggestions on how to debug what's causing this?

...GISTER,REFER,NOTIFY,CANCEL,PRACK,OPTIONS,INFO,SUBSCRIBE,UPDATE Content-Type: multipart/mixed ;boundary=unique-boundary-1 Content-Length: 645 --unique-boundary-1 Content-Type: application/SDP v=0 o=- 95 1 IN IP4 192.168.45.194 s=- t=0 0 m=audio 5234 RTP/AVP 0 8 c=IN IP4 192.168.45.197 a=ptime:20 a=maxptime:20 a=sendrecv --unique-boundary-1 Content-Type: application/x-nt-mcdn-frag-hex ;version=sse-4.00.31 ;base=x2611 Content-Disposition: signal ;handling=optional 05006702 0107130081900000a2 09090f00e9a083000100e7 1315070011fa0f00a10d02010102020100cc046605123c --unique-boundary-1 Content-Type: applic...

Hello list, this is the communication between an Aastra 5000 PBX and Asterisk, where the Aastra makes a call to Asterisk. For some reason, Asterisk responds with 401-Unauthorized and I don't know why. Calls go well with Panasonic PBX, Avaya PBX, Alcatel-Lucent PBX but NOT with this Aastra. A1.A1.A1.A1 = IP-address Asterisk PBX AS.AS.AS.AS = IP-address Aastra PBX Aastra PBX makes a call

...MAC_SHA1_80 inline:Gr23SpFGDiukOKyrrfAauWssQ+3pYjD0jwkK9hOo a=rtpmap:111 opus/48000/2 a=fmtp:111 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=maxptime:60 a=ssrc:2003565451 cname:0Cqf6EiGG5oFoWF5 a=ssrc:2003565451 msid:dXVhxyOSxULu3iClZayhTeEBzH2voboiJJ28 8ae66e18-c0ba-4738-a35c-130a3de87f8f a=ssrc:2003565451 mslabel:dXVhxyOSxULu3iClZayhTeEBzH2voboiJJ28 a=ssrc:2003565451 label:8ae66e18-c0ba-4738-a35c-130a3de87f8f <--- Transmitting SIP respons...

...Attribute Fieldname: fmtp Media Format: 101 [telephone-event] Media format specific parameters: 0-16 Media Attribute (a): ptime:20 Media Attribute Fieldname: ptime Media Attribute Value: 20 Media Attribute (a): maxptime:150 Media Attribute Fieldname: maxptime Media Attribute Value: 150 Media Attribute (a): sendrecv Note that in the SDP part it claims the Owner/Creator (o=) to be 192.168.20.238 which is the main IP address of the box (eth0), but not the one where Asteris...

When I get a SIP call, I get the following error: *CLI> NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is 'multipart/mixed;boundary="unique-boundary-1"', not 'application/sdp' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 5147771111, 1) == Spawn extension (incoming,

...REFER, REGISTER Content-Type: application/sdp Content-Length: 266 v=0 o=- 3690874445 3690874447 IN IP4 222.222.222.22 s=ruVoIP.net PBX c=IN IP4 222.222.222.22 t=0 0 m=audio 25094 RTP/AVP 8 0 96 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 a=ptime:20 a=maxptime:150 a=sendrecv [2016-12-16 13:53:05] VERBOSE[6631] res_pjsip_logger.c: <--- Received SIP request (812 bytes) from UDP:11.111.11.11:5060 ---> UPDATE sip:222.222.222.22:5060 SIP/2.0 Via: SIP/2.0/UDP 11.111.11.11:5060;rport;branch=z9hG4bKPjL4elACGx8R4357HYMDC-eUWi5f5peNYk Max-Forwards: 70 From...

...RISK:5060> Content-Type: application/sdp Require: timer Content-Length: 264 v=0 o=root 227409966 227409966 IN IP4 ASTERISK s=Asterisk PBX 13.14.1~dfsg-2+deb9u4 c=IN IP4 ASTERISK t=0 0 m=audio 13948 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=sendrecv <------------> And that's good to, we've sent the OK for the INVITE saying that we want alaw. <--- SIP read from UDP:SUPPLIER:5060 ---> ACKsip:LOCAL at ASTERISK:5060 SIP/2.0 Via: SIP/2.0/UDP SUPPLIER:5060;branch=z9hG4bK02B5bc037285f864da9 From:<sip...

...REGISTER, MESSAGE, REFER^M Supported: 100rel, timer, replaces, norefersub^M Content-Type: application/sdp^M Content-Length: 179^M ^M v=0^M o=- 32730859 32730861 IN IP4 xxx.xxx.xxx.xxx^M s=Asterisk^M c=IN IP4 xxx.xxx.xxx.xxx^M t=0 0^M m=audio 19384 RTP/AVP 0^M a=rtpmap:0 PCMU/8000^M a=ptime:20^M a=maxptime:150^M a=sendrecv^M ACK sip:xxx.xxx.xxx.xxx:5060 SIP/2.0^M Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5063;branch=z9hG4bK-c38362b^M From: "1004" <sip:1004 at xxx.xxx.xxx.xxx>;tag=79e7940882a792ao2^M To: <sip:333 at xxx.xxx.xxx.xxx>;tag=96156bd7-9e8e-4077-b6e4-f3eb12e39069^M Call-ID: 31...

...; Media Format: 101 [telephone-event] > Media format specific parameters: 0-16 > Media Attribute (a): ptime:20 > Media Attribute Fieldname: ptime > Media Attribute Value: 20 > Media Attribute (a): maxptime:150 > Media Attribute Fieldname: maxptime > Media Attribute Value: 150 > Media Attribute (a): sendrecv > > Note that in the SDP part it claims the Owner/Creator (o=) to be 192.168.20.238 which is the main IP address of the box (eth0), but n...