search for: avp

...eq: 101 INVITE Max-Forwards: 6 Timestamp: 1066825189 Contact: <sip:52880472@200.61.32.142:5060;user=phone> Expires: 180 Content-Type: application/sdp Content-Length: 166 v=0 o=CiscoSystemsSIP-GW-UserAgent 624 4121 IN IP4 200.61.32.142 s=SIP Call c=IN IP4 200.61.32.142 t=0 0 m=audio 20476 RTP/AVP 8 0 18 65535 65535 65535 4 65535 Oct 22 09:19:49.930: Received: SIP/2.0 100 Trying Via: SIP/2.0/UDP 200.61.32.142:5060 From: "52880472" <sip:52880472@200.61.32.142> To: <sip:2003@200.61.32.238;user=phone;phone-context=unknown>;tag=as12560d82 Call-ID: DDEA1FF7-3C011D8-BEBEADB...

...P 172.16.1.54:5060;branch=z9hG4bK3f113fc0c05ec1deece622bd0ed4a521 Max-Forwards: 70 Contact: "3002" <sip:<gateway1>:5060;transport=udp> Content-Type: application/sdp Content-Length: 148 v=0 o=par 0 0 IN IP4 <gateway1> s=- c=IN IP4 <gateway1> t=0 0 m=audio 22224 RTP/AVP 0 3 4 5 6 8 15 18 m=video 22222 RTP/AVP 26 34 31 10 headers, 7 lines Using latest request as basis request Sending to 172.16.1.54 : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found audio format ULAW Found audio format UNKN Found audio format UNKN Found audio format ALAW Found a...

...very hard time with > this > too since, believe it or not, for such a ubiquitous open-source package, > it did not offer a straightforward way of making SQL database dips without > relying on the particular schema of some module or another. > > Ultimately, I found that the 'avp' module has a little function called > avp_db_query() which can extract query results and stick them in > individual > AVP values. This function may have just been put there for kicks, since > what AVP "really" implements is some transparent way of storing key/value >...

...ideo?mail is not stored (audio is through). Both the client?use H.264 codec with following sdp information: Android Based Client SDP Parameters v=0 o=- 1325786904 1325786904 IN IP4 172.16.130.47 s=Polycom RealPresence c=IN IP4 172.16.130.47 b=AS:1920 t=0 0 a=sendrecv m=audio 3230 RTP/AVP 118 115 114 113 0 8 119 a=rtpmap:118 SIRENLPR/48000 a=fmtp:118 bitrate=64000 a=rtpmap:115 G7221/32000 a=fmtp:115 bitrate=48000 a=rtpmap:114 G7221/32000 a=fmtp:114 bitrate=32000 a=rtpmap:113 G7221/32000 a=fmtp:113 bitrate=24000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:119 teleph...

Hi, I am using codec g729 on two asterisk machines, but when call is forwarded from asterisk server 1 (32bit) to server 2(64bit). server 1 outputs following error and there is no audio. Also the IVRs being played have choppy voice. "Insufficient information for SDP (m = 'audio 0000 RTP/AVP 18 127', c = '')" It is running fine when codec gsm is in RTP traffic. Also I have another server 3 which is also running g729, call from server 3 to server 2 is established but still choppy voice. Earlier I integrated server 3 to server 1 and it was a smooth run. Any idea what...

While trying to use direct_media I'm seeing RTP payload mismatches after succesful reinvites. Initial INVITE from endpoint A to asterisk has rfc4733 DMTF m=audio 35648 RTP/AVP 9 8 111 96 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-16 >From asterisk to upstream U: m=audio 14338 RTP/AVP 9 8 111 18 0 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 So the payload types in the RTP streams from A and to U differ. This works fine when asterisk is relaying media. Wi...

..._line(resp, m_video->str); add_line(resp, a_video->str); add_line(resp, hold); /* Repeat hold for the video stream */ } else if (p->offered_media[SDP_VIDEO].offered) { snprintf(dummy_answer, sizeof(dummy_answer), "m=video 0 RTP/AVP\ %s\r\n", p->offered_media[SDP_VIDEO].text); add_line(resp, dummy_answer); But "len", which was used to set Content-Length, isn't updated to onclide that dummy. Doesn't it need to be? I think this may be a problem with a connection to my Polycom VSX.

...pposed to have started. When debugging with tcpdump, I have seen that all the successful calls have SDP negotiation that look like this: (inside INVITE request 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 IP...

I installed avp 2010 and wanted to have some fun in mp, but when starting the game, I had no sound at all... The weird thing is that in movies and such, there is sound. It's muted ingame and in menus only... terminel output (shortened): Code: fixme:d3d9:D3DPERF_SetOptions (0x1) : stub fixme:thread:SetThreadI...

if i do an rsync to the current directory, how come the parent directory isn't created? I normally just get my files spewed into the current directory, eventhough I am trying to transfer a directory. Something like this rsync -avp host::module/ . -- Jason G Helfman Network Administrator BizRate.com 310.754.1264 desk 310.466.2319 cell Fingerprint: DA13 C109 072B CC12 B568 8D84 E9A2 6A7D C479 BCFB...

...90@x.x.x.x>;tag=839C4028-E83 Call-ID: 8780-3280658280-475041@x.x.x.x CSeq: 1 INVITE Via: SIP/2.0/UDP x.x.x.x:5060 Contact: sip:002125095690@x.x.x.x:5060 Content-Type: application/sdp Content-Length: 267 v=0 o=labis01 1234 5795 IN IP4 x.x.x.x s=sip call c=IN IP4 x.x.x.x t=0 0 m=audio 17184 RTP/AVP 18 0 4 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:4 G723/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 10 headers, 12 lines Using latest request as basis request Sending to x.x.x.x : 5060 (non-NAT) Found audio format UNKN Found audio format UNKN Found...

I receive an INVITE/SDP containing: m=audio 11310 RTP/AVP 3 0 101 which I interpret as gsm, ulaw, rfc2833. and I reply with an OK/SDP containing: m=audio 15884 RTP/AVP 0 3 101 which I interpret as ulaw, gsm, rfc2833. How can I tell which codec was actually used for the call? -- Thanks in advance, --------------------------------------------------...

...q: 101 INVITE Max-Forwards: 6 Timestamp: 1064519388 Contact: <sip:52880472@172.16.254.96:5060;user=phone> Expires: 180 Content-Type: application/sdp Content-Length: 167 v=0 o=CiscoSystemsSIP-GW-UserAgent 8010 6925 IN IP4 172.16.254.96 s=SIP Call c=IN IP4 172.16.254.96 t=0 0 m=audio 20388 RTP/AVP 8 0 18 65535 65535 65535 4 65535 15 headers, 6 lines Using latest request as basis request Sending to 172.16.254.96 : 5060 (non-NAT) Found audio format ALAW Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UNKN Found audio format UL...

...the following example - would be very helpful in such cases: Description of "488 Not acceptable here" - Could be caused by codec problems, when codec negotiation failed. You can check if the negotiation failed by .... - Can be paused by a phone offering encryption, but only offering RTP/AVP instead of RTP/SAVP profile. Check if the sip log contains a crypto line and only RTP/AVP, if yes, change the phone settings from RTP/AVP to RTP/SAVP or disable RTP encryption in the phone's settings. [Even better: Besides throwing the error message also add the reason for it, at least in the A...

...EFER User-Agent: PolycomSoundPointIP-SPIP_601-UA/2.0.3.0127 Supported: 100rel,replaces Allow-Events: talk,hold,conference Max-Forwards: 70 Content-Type: application/sdp Content-Length: 245 v=0 o=- 978307756 978307756 IN IP4 10.253.4.50 s=Polycom IP Phone c=IN IP4 10.253.4.50 t=0 0 m=audio 2228 RTP/AVP 0 18 8 101 a=sendrecv a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 --- (14 headers 11 lines) --- Using INVITE request as basis request - a857d7ac-36f29d46-4d6ef889@10.253.4.50 Sending to 10.253.4.50 : 5060 (NAT) Reliably Transmitting (no NAT)...

...e installed Windows NT 4.0. These two PC's are connected to the same network. I've configured Samba to share the home directories on the Linux System. I'm able to reach my own homedir from NT. Within NT I've assigned this share to drive L. I'm also running AntiVirus Pro 3.0 (www.avp.com) with background monitoring enabled. This software installs these services: - F-SECURE AVP - F-SECURE Filter - F-SECURE Gatekeeper - F-SECURE Recognizer I'm also using Netscape Communicator PRO 4.7. Recently I've moved my Netscape-profile to drive L (my homedir). Then I started Netscap...

...192.168.1.5> >;tag=f543a140 Call-ID: MjFlODY2MmE0OTBmMzE2ZDJhMWEzN2ZmYzUwM2ZkODg. CSeq: 2 INVITE Content-Type: application/sdp User-Agent: X-Lite release 1104o stamp 56125 Content-Length: 261 v=0 o=- 8 2 IN IP4 192.168.1.4 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.4 t=0 0 m=audio 50420 RTP/AVP 107 0 8 101 <--- Transmitting (no NAT) to 192.168.1.4:18341 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.4:18341 ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;received=192.168.1.4;rport=18341 From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5&...

Using 3.0.5 I can't send a file with ':' in its name in the current directory without prefixing it with ./ delos% touch file:ext delos% rsync -avP file:ext sylla:/home/ldm The source and destination cannot both be remote. rsync error: syntax or usage error (code 1) at main.c(1154) [receiver=3.0.5] delos:~% rsync -avP "file:ext" sylla:/home/ldm The source and destination cannot both be remote. rsync error: syntax or usage error (cod...

Dear all, I have Asterisk 1.2.13 running OK with Twinkle clients, they can talk very well using SIP. I have a Jabber server running OK and the clients use PSI client for chat succesfully. Now I'm using Wengophone 2.1.1 in order to unify voip+IM services. The users can logon OK in SIP and Jabber, they get the online status presence, but they CAN'T talk and chat among them.

Hi mates, times ago I've taken the whim to buy aliens versus predator classic 2000 from Steam. It's quite fine, the game works good except the mouse. mouse axes don't work, mouse don't moves. I don't know what to do to fix it, I've tried to set up MouseWarpOverride to force/disable/enable but nothing happens. I've opened a report on bugzilla too, but I'm not having