search for: sipua

When I use an analog phone connected to a Sipura SPA-2000 it takes about 3-4 seconds before the number is actually dialled. Very annoying especially if you are connecting an intercom to it. Can I change this behaviour and do I need to look at * config or the config of the SPA-2000? Thanks!

...<sip:1002@192.168.1.102:5060;transport=udp> Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Supported: replaces,join,norefersub Content-Length: 274 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 1914 0 IN IP4 192.168.1.102 s=SIP Call t=0 0 m=audio 25584 RTP/AVP 0 8 18 101 c=IN IP4 192.168.1.102 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/0 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (16 headers 13 lines)--- Using INVITE request as ba...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I was just measuring the bitrates of a couple of codecs via iax. I'm getting much higher numbers than expected, so maybe I'm doing something wrong? Measured with iptraf, values displayed are: codec: measured bitrate (bitrate according codec definition) gsm: 52 kbps (13 kpbs) alaw: 154 kbps (?) speex: 57 kpbs (24 kpbs) Seems a little

...em that is alien to me and obvious for some of you :). I have asterisk setup with few sip clients(using linphonec). In a proper context, I have mentioned extensions 107 as simputer@X.X.X.X (x.x.x.x=asterisk server ip) Asterisk Sever-------------------------simputer(sip ua) I can make calls from sipua to asterisk but not reverse way. I get the following display on asterisk terminal. Could anyone of you please tell em why im not able to receive calls on linphonec ?n --------------------- *CLI> -- Executing Dial("SIP/clienta-30c9", "SIP/simputer|20|tr") in new stack...

Hello All, I have a problem that is alien to me and obvious for some of you :). I have asterisk setup with few sip clients. In a proper context, I have mentioned extensions 107 as simputer@bogus.com Asterisk Server-------------------------simputer(sip ua) I can make calls from sipua to asterisk but not reverse way. I get the following display on asterisk terminal --------------------- *CLI> -- Executing Dial("SIP/clienta-30c9", "SIP/simputer|20|tr") in new stack -- Called simputer Dec 28 12:00:05 WARNING[-1116775504]: chan_sip.c:665 retrans_pkt:...

.... Is there any way that could cause this error though? It appears the encryption details are sufficient and do not otherwise differ between Firefox 24 and 26: --- ff-24.txt 2014-08-25 15:02:20.452383599 +0200 +++ ff-26.txt 2014-08-25 15:01:42.472346613 +0200 @@ -1,12 +1,12 @@ v=0 -o=Mozilla-SIPUA-24.7.0 14737 0 IN IP4 0.0.0.0 +o=Mozilla-SIPUA-26.0 18111 0 IN IP4 0.0.0.0 s=SIP Call t=0 0 -a=ice-ufrag:301212e4 -a=ice-pwd:d7430f468514f1f2d326d3c944691fbf -a=fingerprint:sha-256 E2:53:6A:FA:6D:E2:3F:7E:24:82:0F:E3:27:34:D1:CC:50:31:42:82:5F:DF:34:9A:4F:42:D1:6D:B7:DB:5C:43 -m=audio 54908 UDP/T...

...CSeq: 101 INVITE User-Agent: CSCO/5 Contact: <sip:5285@192.168.1.84:5060> Expires: 180 Content-Type: application/sdp Content-Length: 246 Accept: application/sdp Remote-Party-ID: "5285" <sip:5285@192.168.1.84>;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 9972 27311 IN IP4 192.168.1.84 s=SIP Call c=IN IP4 192.168.1.84 t=0 0 m=audio 31790 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 14 headers, 11 lines Using latest request as basis request Sending to 192.168.1.8...

...-Agent: CSCO/4 Contact: <sip:3015321510@128.151.224.33:5060> Expires: 180 Content-Type: application/sdp Content-Length: 247 Accept: application/sdp Remote-Party-ID: "3015321510" <sip:3015321510@128.151.224.33>;party=calling;id-type=subscriber;privacy=off;screen=no v=0 o=Cisco-SIPUA 15975 21108 IN IP4 128.151.224.33 s=SIP Call c=IN IP4 128.151.224.33 t=0 0 m=audio 19364 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 14 headers, 11 lines Using latest request as basis request Sending to 128.15...

...,OPTIONS,REFER,REGISTER,UPDATE Remote-Party-ID: "Cisco 7940" <sip:215@10.0.10.5>;party=calling;id-type=subscriber;privacy=off;screen=yes Supported: replaces,join,norefersub Content-Length: 274 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 19361 0 IN IP4 10.0.10.136 s=SIP Call t=0 0 m=audio 16946 RTP/AVP 8 0 18 101 c=IN IP4 10.0.10.136 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv 2007-04-12 15:39:56 DEBUG[11050]: chan_sip.c:3442 parse...

...webrtc avpf=yes ; needed for webrtc context=default encryption=yes dtlsenable=yes dtlsverify=no dtlsrekey=60 dtlscafile=/opt/asterisk/keys/ca.crt dtlscertfile=/opt/asterisk/keys/asterisk.pem dtlssetup=actpass insecure=invite Here is the SDP offered by Nightly: v=0 o=Mozilla-SIPUA-24.0a1 25687 1 IN IP4 0.0.0.0 s=Doubango Telecom - firefox t=0 0 a=ice-ufrag:7194cbcc a=ice-pwd:e57c14491015e529b84c5a6baf6d7b67 a=fingerprint:sha-256 48:3E:0C:59:BA:EB:6C:F9:5D:65:BF:08:54:63:C3:EA:AF:A9:60:9D:39:47:A5:41:6B:E1:A8:EB:7C:06:BE:D4 m=audio 62583 UDP/TLS/RTP/SAVPF 109 0 8 101 c=IN IP4...

...<sip:111 at 192.168.1.61:5060;transport=udp> Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE Allow-Events: kpml,dialog Content-Length: 322 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 626 0 IN IP4 192.168.1.61 s=SIP Call t=0 0 m=audio 30354 RTP/AVP 0 8 18 116 101 c=IN IP4 192.168.1.61 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv <--- T...

Oh god it works ! to switch cisco to upd I used config: <transportLayerProtocol>2</transportLayerProtocol> with udp it works well, thanks for your help :) > On 24 Feb 2015, at 17:02, Joshua Colp <jcolp at digium.com> wrote: > > If you use UDP with force_rport=no it'll work. > If you use TCP then set rewrite_contact=yes so it'll reuse the established TCP

...domain.com;user=phone> Call-ID: 00078599-323d0005-0dc35ec5-5770d68a@82.33.200.166 Date: Fri, 21 Jan 2005 11:13:19 GMT CSeq: 101 INVITE User-Agent: CSCO/6 Contact: <sip:30@82.33.200.166:5060> Expires: 180 Content-Type: application/sdp Content-Length: 247 Accept: application/sdp v=0 o=Cisco-SIPUA 286 22351 IN IP4 82.33.200.166 s=SIP Call c=IN IP4 82.33.200.166 t=0 0 m=audio 29280 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 13 headers, 11 lines Using latest request as basis request Sending to 82.33.200....

...User-Agent: CSCO/6 Contact: <sip:nmartin@172.31.30.7:5060> Expires: 180 Content-Type: application/sdp Content-Length: 244 Accept: application/sdp Remote-Party-ID: "105 - Nik Martin" <sip:nmartin@172.31.30.7>;party=calling;id-type=subscriber;privacy=off;scree n=no v=0 o=Cisco-SIPUA 24482 2915 IN IP4 172.31.30.7 s=SIP Call c=IN IP4 172.31.30.7 t=0 0 m=audio 26676 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 14 headers, 11 lines Using latest request as basis request Sending to 172.31.30.7 :...

...port for 3 common codecs: a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 Now of course each device specified these 3 codecs in a different order. Under normal circumstances I feel this call should complete why is * claiming a codec mismatch? - Dustin - From phone v=0 o=Cisco-SIPUA 5892 12461 IN IP4 192.168.68.12 s=SIP Call c=IN IP4 192.168.68.12 t=0 0 m=audio 18114 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 Sent to remote server by * v=0 o=root 4205 4205 IN IP4 X.X.X.X s=session c=IN...

...b96b81e To: <sip:1phoneiamcalling@xxx.xxx.xxx.xxx> Call-ID: 00036b09-607e0047-3dc1c568-1d31a410@ipoftphone CSeq: 101 INVITE User-Agent: CSCO/6 Contact: <sip:phonesoftphone@ipoftphone:5060> Expires: 180 Content-Type: application/sdp Content-Length: 249 Accept: application/sdp v=0 o=Cisco-SIPUA 28416 11732 IN IP4 ipoftphone s=SIP Call c=IN IP4 ipoftphone t=0 0 m=audio 26298 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 13 headers, 11 lines Using latest request as basis request Sending to xxx.xxx.xxx.xxx...

...sco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-6.0.2,X-cisco-xsi-8.5.1 Allow-Events: kpml,dialog Content-Length: 685 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 27778 0 IN IP4 10.168.154.71 s=SIP Call t=0 0 m=audio 10032 RTP/AVP 0 8 18 102 9 116 101 c=IN IP4 10.168.154.71 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:102 L16/16000 a=rtpmap:9 G722/8000 a=rtpmap:116 iLBC/8000 a=fmtp:116 mode=20 a=rtpmap:101 tele...

...43a6c To: <sip:94732771@yyy.yy.113.56;user=phone> Call-ID: 0003e362-fc0a0042-22647abb-4243e54a@xxx.xxx.226.166 CSeq: 101 INVITE User-Agent: CSCO/4 Contact: <sip:TOMS@xxx.xxx.226.166:5060> Expires: 180 Content-Type: application/sdp Content-Length: 250 Accept: application/sdp v=0 o=Cisco-SIPUA 20902 8155 IN IP4 xxx.xxx.226.166 s=SIP Call c=IN IP4 xxx.xxx.226.166 t=0 0 m=audio 27632 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 12 headers, 11 lines Using latest request as basis request Sending to xxx.x...

I have a group of users configured as extensions in *.These users are registered with a SIP Proxy Server and can receive calls very well. The problem happens when any user tries to make an outbound call. The proxy replies with a "401 Unauthorized" and * don't try another INVITE including credentials. Here is part of the content of sip.conf. [general] port = 5061 bindaddr = *.IP

...<sip:304 at 192.168.96.18:5060;user=phone;transport=udp> Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE Supported: replaces,join,norefersub Content-Length: 278 Content-Type: application/sdp Content-Disposition: session;handling=optional v=0 o=Cisco-SIPUA 26847 2 IN IP4 192.168.96.18 s=SIP Call t=0 0 m=audio 26612 RTP/AVP 0 8 18 101 c=IN IP4 192.168.96.18 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly <-------------> --- (16 headers 13 lines) ---...