search for: d8754z

Hi all, Asterisk is great but I'm having issues with setting up realtime for our call center, which is needed for login integration with the rest of our applications (telephonists' web interface, etc.). I have reviewed a large number of previous posts to the mailing list and the voip-info wiki to no avail. Setup is as follows: Linux 2.6.23 (gentoo) / AMD Athlon(tm) 64 Processor 3000+ /

...; 12345,1,Dial(SIP,giusy*) Below the output of SIP debug of IP caller (192.168.1.116) in asterisk *dhcppc0*CLI> <--- SIP read from 192.168.1.116:14862 ---> INVITE sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100> SIP/2.0 Via: SIP/2.0/UDP 192.168.1.116:14862 ;branch=z9hG4bK-d8754z-0254e549a042446a-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:gianca at 192.168.1.116:14862> To: "12345"<sip:12345 at 192.168.1.100 <sip%3A12345 at 192.168.1.100>> From: "gianca"<sip:gianca at 192.168.1.100 <sip%3Agianca at 192.168.1.100> >;tag=...

...received 181.64.96.122: Is my home IP 190.12.68.20 or central.cipher.pe: is office IP 206.15.148.221: Broadvoice Server <--- SIP read from UDP:181.64.96.122:11037 ---> INVITE sip:90018006273999 at central.cipher.pe SIP/2.0 Via: SIP/2.0/UDP 192.168.7.33:19116;branch=z9hG4bK-d8754z-81993d517bc9b121-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:488 at 181.64.96.122:11037> To: "90018006273999"<sip:90018006273999 at central.cipher.pe> From: "488"<sip:488 at central.cipher.pe>;tag=93cce179 Call-ID: ZDk2MDVkY2RhMTE2...

...s there any way to setup rtp directly between 2 sip clients, no need to go through asterisk server here is my debug log: <--- SIP read from UDP://192.168.1.4:18341 ---> INVITE sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5> SIP/2.0 Via: SIP/2.0/UDP 192.168.1.4:18341 ;branch=z9hG4bK-d8754z-da695e1f167deb68-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:test at 192.168.1.4:18341> To: "1000"<sip:1000 at 192.168.1.5 <sip%3A1000 at 192.168.1.5>> From: "Do Nguyen Ha"<sip:test at 192.168.1.5 <sip%3Atest at 192.168.1.5> >;tag=f543a140 Call...

...e xlite but still does not work. I have traces of xlite for the invite and register this done to see if someone can help me to use this line with my asterisk. These are the traces of my Xllite REGISTER sip:Xlite release 1100l stamp 49022 SIP/2.0 Via: SIP/2.0/UDP 10.0.0.221:22818;branch=z9hG4bK-d8754z-e322ee549824f666-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:8887776666 at 10.0.0.221:22818;rinstance=570ac597afa82c9a> To: "8887776666"<sip:8887776666 at Xlite release 1100l stamp 49022> From: "8887776666"<sip:8887776666 at Xlite release 1100l stamp 49022&...

...t's it... Other Snom phones with SIP-accounts go very well, but at this location the registration fails. Another remark : when using a Zoiper softphone, the registration goes very well : REGISTER sip:sip.domain.tld;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 192.168.114.20:5060;branch=z9hG4bK-d8754z-fab4a5effbf90a05-1---d8754z- Max-Forwards: 70 Contact: <sip:test3 at public_ip:51363;rinstance=b6fd38105c91b9bf;transport=UDP> To: <sip:test3 at sip.domain.tld;transport=UDP> From: <sip:test3 at sip.domain.tld;transport=UDP>;tag=db1a5018 Call-ID: NzBlZDMyN2U0YTEzZDk4Y2M2N2NmNzMxY...

...ect. So what can I do to find out where went wrong on this sip trunk? Thanks. Jian Hers is the debug out put: ============================ <--- SIP read from 208.65.xxx.xxx:5060 ---> INVITE sip:160428xxxxx at 192.168.1.83:5060 SIP/2.0 Via: SIP/2.0/UDP 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-ad239907e0915d0b-1---d8754z-;rport Via: SIP/2.0/UDP 208.65.xxx.xxx:5061;branch=z9hG4bK-pcerhxpz5hr4addh;rport=5061 Max-Forwards: 69 Record-Route: <sip:208.65.xxx.xxx;lr> Contact: "Anonymous"<sip:208.65.xxx.xxx:5061> To: <sip:160428xxxxx at 208.65.xxx.xxx:5060> From: &qu...

Hi! AFAIS the incoming SUBSCRIBE is handled in the same context as INVITE - but how should I handle the SUBSCRIBE in the context? thanks klaus SUBSCRIBE sip:u+431234567 at foobar.at:5160 SIP/2.0 Via: SIP/2.0/UDP 192.168.2.82:39982;branch=z9hG4bK-d8754z-3116e1207913aa4e-1---d8754z-;rport Max-Forwards: 70 Contact: <sip:u+431234567 at 11.111.11.11:39982> To: "schlopy"<sip:u+431234567 at foobar.at:5160> From: "schlopy"<sip:u+431234567 at foobar.at:5160>;tag=376b6b2e Call-ID: ZDY1MmExZDdlNGE0MGI0NzgxZGQxMjA5YWNmMT...

...of sip trunk stop working after the upgrade. Here is the sip debug: --------------------------------------------------------------------------- <--- SIP read from 208.65.xxx.xxx:5060 ---> INVITE sip:1778xxxxxxx at 10.11.22.77:5060 SIP/2.0 Via: SIP/2.0/UDP 208.65.xxx.xxx:5060;branch=z9hG4bK-d8754z-d9175178645e9146-1---d8754z-;rport Via: SIP/2.0/UDP 208.65.xxx.xxx:5061;branch=z9hG4bK-uhhmj2ir4ew6cn4p;rport=5061 Max-Forwards: 69 Record-Route: <sip:208.65.xxx.xxx;lr> Contact: "Anonymous"<sip:208.65.xxx.xxx:5061> To: <sip:1778xxxxxxx at 208.65.xxx.xxx:5060> From: &lt...

...ol (REGISTER) Request-Line: REGISTER sip:1.2.3.4;transport=TCP SIP/2.0 Method: REGISTER Request-URI: sip:1.2.3.4;transport=TCP Request-URI Host Part: 1.2.3.4 [Resent Packet: False] Message Header Via: SIP/2.0/TCP 192.168.1.15:47053 ;branch=z9hG4bK-d8754z-5e3d9f441f1de1d3-1---d8754z-;rport;transport=TCP Transport: TCP Sent-by Address: 192.168.1.15 Sent-by port: 47053 Branch: z9hG4bK-d8754z-5e3d9f441f1de1d3-1---d8754z- RPort: rport transport=TCP Max-Forwards: 70 C...

Thank you for your note, Scott. I set rewrite_contact=yes for both contacts, and I also had to do remove_existing=yes because I had to remove the existing contact information (max_contacts = 1 was preventing new contact information) using pjsip qualify demo-alice etc., after which the right IP addresses showed in pjsip show endpoints. Anyway, it works as expected now, I think. My pjsip.conf is

Thanks for the mighty quick response, Joshua! I am using Zoiper on Linux softclient: REGISTER sip:<ipAddr>;transport=TCP SIP/2.0 Changed the port back to 5060. On Mon, Feb 15, 2016 at 1:40 PM, Joshua Colp <jcolp at digium.com> wrote: > Sonny Rajagopalan wrote: > > <snip> > > > *CLI> pjsip set logger on >> PJSIP Logging enabled >> [Feb 15

...thinks it?s connected. Logs say ?no reply to our critical packet?. tcpdump shows that the packet does arrive at the destination. sip set debug shows this is what the packet contains: Retransmitting #6 (NAT) to 77.239.189.223:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 77.239.189.223;branch=z9hG4bK-d8754z-db899ced94cc7fd3-1---d8754z-;received=77.239.189.223 From: "Roma"<sip:roma at qwertty.com;transport=UDP>;tag=01785d5e To: <sip:echo at qwertty.com;transport=UDP>;tag=as068592d2 Call-ID: ZTkzNjYxNzZmOWMzY2ZhOTdjMWIwYTEwZTYxZmUyZTY. CSeq: 2 INVITE User-Agent: Asterisk PBX Allow:...

...terisk are at 1.1.1.1) INVITE sip:660 at testers.com;transport=UDP SIP/2.0 Record-Route: <sip:1.1.1.1;lr=on;ftag=fd070807> Via: SIP/2.0/UDP 1.1.1.1;branch=z9hG4bKf6e9.339dda0648d95af665c91db701754d98.0 Via: SIP/2.0/UDP 2.2.2.2:37730 ;rport=37730;branch=z9hG4bK-d8754z-7f27c9fc35574abb-1---d8754z- Max-Forwards: 16 Contact: <sip:700 at 2.2.2.2:37730;transport=UDP> To: <sip:660 at testers.com;transport=UDP> From: <sip:700 at testers.com;transport=UDP>;tag=fd070807 Call-ID: ZDc0YjU1ZjNmMWI5YjUyYzY0YWNjN2NjN2N...

...not being set up, but all the codec support is there. Here's a log for the SIP request from 192.168.1.50: <--- Received SIP request (1229 bytes) from UDP:192.168.1.50:64009 ---> INVITE sip:6002 at 192.168.1.139;transport=UDP SIP/2.0 Via: SIP/2.0/UDP 146.115.163.234:64009 ;branch=z9hG4bK-d8754z-5803642ad92cbd00-1---d8754z- Max-Forwards: 70 Contact: <sip:demo-alice at 146.115.163.234:64009;transport=UDP> To: <sip:6002 at 192.168.1.139;transport=UDP> From: <sip:demo-alice at 192.168.1.139;transport=UDP>;tag=b661670b Call-ID: YmZiODE4Yzc2NmJmNzY5NDhkM2Y2ZDNhM2U4NTZmZmE. CSe...

...60 > PU.BL.IC.IP:5070: SIP, length: 1046 INVITE sip:660 at testers.com;transport=UDP SIP/2.0 Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177> Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bKd7b.ca8b6ac6a82d605cf658af0fea7c9e86.0 Via: SIP/2.0/UDP AST.ER.ISK.IP:38699;rport=38699;branch=z9hG4bK-d8754z-bd00e9fd46368417-1---d8754z- Max-Forwards: 69 Contact: <sip:771 at AST.ER.ISK.IP:38699;transport=UDP> To: <sip:660 at testers.com;transport=UDP> From: "771"<sip:771 at testers.com;transport=UDP>;tag=41030177 Call-ID: YWYwMjMwMmZlODEwM2MwODdjZWJmYjc2NjM5ZmIyNzk. CSeq: 2 I...

...> <mailto:sip%3A660 at testers.com>;transport=UDP SIP/2.0 > Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177> > Via: SIP/2.0/UDP > PU.BL.IC.IP;branch=z9hG4bKd7b.ca8b6ac6a82d605cf658af0fea7c9e86.0 > Via: SIP/2.0/UDP > AST.ER.ISK.IP:38699;rport=38699;branch=z9hG4bK-d8754z-bd00e9fd46368417-1---d8754z- > Max-Forwards: 69 > Contact: <sip:771 at AST.ER.ISK.IP:38699;transport=UDP> > To: <sip:660 at testers.com <mailto:sip%3A660 at testers.com>;transport=UDP> > From: "771"<sip:771 at testers.com > <mailto:sip%3A771 at test...

...ip:50 at 192.168.1.109>' failed for ' 192.168.1.80:5062' - Wrong password even though on 1.6 everything was OK here is part of debug messages: ---cut--- <--- Transmitting (NAT) to 192.168.1.50:5062 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.50:5062;branch=z9hG4bK-d8754z-d7545057f425cd49-1---d8754z-;received=192.168.1.50;rport=5062 From: "51"<sip:51 at 192.168.1.109:5062> <sip:51 at 192.168.1.109:5062>;tag=172a701e To: "51"<sip:51 at 192.168.1.109:5062> <sip:51 at 192.168.1.109:5062>;tag=as6773fc96 Call-ID: ODlkNDYyMmY...

...0000002' [Jul 9 21:39:29] DEBUG[47716][C-00000002]: chan_pjsip.c:1578 hangup_cause2sip: AST hangup cause 0 (no match found in PJSIP) <--- Transmitting SIP response (369 bytes) to UDP:1.1.1.1:49260 ---> SIP/2.0 603 Decline v: SIP/2.0/UDP 1.1.1.1:49260;rport;received=1.1.1.1;branch=z9hG4bK-d8754z-22994e127365d474-1---d8754z- i: MmFjNDM4NDc2NmFhZWNiYTU2MDQ1YmNjNGVmYmMyOTY f: "9544447408" <sip:9544447408 at 8.26.191.189>;tag=82c82c1d t: <sip:17274428141 at 8.26.191.189>;tag=09f3a67a-f457-46d1-8d16-243478ac3859 CSeq: 1 INVITE Reason: Q.850;cause=0 l: 0

...RTP is at port 192.168.0.73:40958 Looking for 702 in from-internal (domain ABC.dyndns.org) list_route: hop: <sip:701 at 123.456.789.000:37587> acerdebian*CLI> <--- Transmitting (NAT) to 123.456.789.000:28127 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.0.73:15158 ;branch=z9hG4bK-d8754z-0a540c5d3439c271-1---d8754z-;received=123.456.789.000;rport=28127 From: "me"<sip:701 at ABC.dyndns.org <sip%3A701 at ABC.dyndns.org>>;tag=3c08d834 To: "702"<sip:702 at ABC.dyndns.org <sip%3A702 at ABC.dyndns.org>> Call-ID: Y2QxNTg4NjE3MTZjNGMzZGM5NzE3YWY4...