search for: atasip

...DP 192.168.0.1:5060;branch=z9hG4bK286113b7;rport From: "123" <sip:123 at 192.168.0.1>;tag=as0cd1aab0 To: <sip:1 at 192.168.0.70:5060;user=phone;transport=udp>;tag=2035093099 Call-ID: 3daa9e730e767bf932a9196a35200e36 at 192.168.0.1 CSeq: 102 INVITE Server: Cisco ATA 186 v3.2.1 atasip (050616A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE Content-Length: 0 <-------------> --- (9 headers 0 lines) --- gw*CLI> <--- SIP read from 192.168.0.70:5060 ---> SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK286113b7;r...

I'm getting some warnings: NOTICE[xxx]: File rtp.c, Line 264 (process_rfc3389): RFC3389 support incomplete. Turn off on client if possible Asterisk Version: CVS-01/06/04-13:50:26 Cisco ATA 186 version: v3.0.0 atasip (Build 031210A) Is this something I should be concerned about? Anyone know how to "turn off" the RFC3389 support on the ata 186? Thanks!

Hi, I?m looking for settings to configure net2phone carrier in my asterisk. I found this configurations, but it?s not work. I don?t known if this configuration is for voice line or voice access account. Anybody can help me, with other configuration? Thanks. ---- *sip.conf* [general] useragent = X-Lite release 1103m register => PHONENUMBER:PASSWORD@sip.net2phone.com [net2phone] type = peer

...3213084003 <sip:3213084003@xxx.xxx.xxx.xxx;user=phone>;tag=2785115378 To: <sip:1001@xxx.xxx.xxx.xxx;user=phone> Call-ID: 1801199744@192.168.200.169 CSeq: 1 INVITE Contact: 3213084003 <sip:3213084003@192.168.200.169:5060;user=phone;transport=udp> User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A) Expires: 300 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 280 Content-Type: application/sdp v=0 o=3213084003 1311 1311 IN IP4 192.168.200.169 s=ATA186 Call c=IN IP4 192.168.200.169 t=0 0 m=audio 16384 RTP/AVP 18 8 0 101 a=rtpmap:18 G729/8000/1 a=fmtp:...

...2.0/UDP 10.44.16.163:5060 From: <sip:4561@82.70.154.145;user=phone>;tag=2361964166 To: <sip:4561@82.70.154.145;user=phone> Call-ID: 1812954233@10.44.16.163 CSeq: 1 REGISTER Contact: <sip:4561@10.44.16.163:5060;user=phone;transport=udp>;expires=120 User-Agent: Cisco ATA 186 v3.1.0 atasip (040211A) Content-Length: 0 --- (9 headers 0 lines)--- Using latest request as basis request Sending to 10.44.16.163 : 5060 (NAT) Transmitting (NAT) to 212.74.112.53:8413: SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.44.16.163:5060;received=212.74.112.53;rport=8413 From: <sip:4561@82.70.154.145;user...

...;sip:xxx9931211@proxy.abccorp.net>;tag=1841513983 Call-ID: 58d4bd4e5e29ff254db520665915ac83@xxx.yyy.77.23 CSeq: 102 INVITE Contact: <sip:xxx9931211@xxx.yyy.165.201:5060;user=phone;transport=udp> Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Server: Cisco ATA 186 v3.0.0 atasip (031210A) Content-Type: application/sdp Content-Length: 205 v=0 o=xxx9931211 18172 18172 IN IP4 xxx.yyy.165.201 s=ATA186 Call c=IN IP4 xxx.yyy.165.201 t=0 0 m=audio 16384 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 --------------------------------------...

I have coded chan_sip.c so that you can have // sip.conf register => username:password@somedomain.com/redirectconfig [redirectconfig] redirect=yes redirecturi=sip:12345@domain1.com redirecturi=sip:34556@domain2.com redirecturi=sip:87877@domain3.com .... so when you receive a call it will redirect to the alternating uri's with a SIP 300 Message. It works with the following sequence,

....18.125:5060;branch=z9hG4bK0b1527ad From: "NafthaChimie" <sip:2000@10.100.18.125>;tag=as49257c3d To: <sip:2001@10.100.18.124:5060;user=phone;transport=udp>;tag=556017164 Call-ID: 256db95b2c935d47464a5d0b7d9f82c4@10.100.18.125 CSeq: 102 INVITE Server: Cisco ATA 186 v3.1.0 atasip (040211A) Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER Content-Length: 0 Warning: Media type not available 10 headers, 0 lines Transmitting: ACK sip:2001@10.100.18.124:5060;user=phone;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.100.18.125:5060;branch=z9hG4bK0b1527ad...

Hi All, Having searched the archives, I can see there has been much discussion at various points regarding capture of caller id information from good old BT. If I understand correctly, it seems that not only do the drivers not currently support it, but my X101P possibly/probably can't do it anyway due to hardware? So, that leaves me with the modem route, which seems more and more unlikely,