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...ersion) with asterisk. however no mention of VoIP protocols was mentioned as the wellgates traditionally supported H.323 but with firmware upgrade has been able to support SIP. (sip debug output begins) Sip read: REGISTER sip:192.168.0.200:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.202:5060;branch=z9hG4bK-0-35c-47a0 Max-Forwards: 70 Supported: replaces User-Agent: FXS_GW (4asipfxs.107a) Contact: <sip:1234@192.168.0.202:5060>;expires=60 From: <sip:1234@192.168.0.200> ;tag=c0a800ca-13c4-0-35c-48a3 To: <sip:1234@192.168.0.200> Call-ID: c0a800ca-13c4-0-334-1c34 CSeq: 1 REGISTER Content...

...ecretary Answered by Secretary ******************************************************************************** Received from udp:209.189.239.106:5060 at 24/8/2004 11:35:01:770 (1163 bytes): NOTIFY sip:9723048720@4.12.220.193:5060;line=t5116uyl SIP/2.0 Via: SIP/2.0/UDP 209.189.239.106:5060;branch=z9hG4bK-05024cf591289707a53e4345ef25f87f.1 Via: SIP/2.0/UDP 4.13.147.200:11636;branch=z9hG4bK-98k96vibydan;rport Record-Route: <sip:abpusa.com:5060;maddr=209.189.239.106> From: <sip:9723048722@abpusa.com;user=phone>;tag=wr4771pry1 To: <sip:9723048720@abpusa.com>;tag=gh50agmbxb Call-ID: 3c...

...I've attached a sanitized sip trace from the snom phone for your perusal. Thanks for any help you can offer. Brian ### START SIP TRACE ### Sent to udp:192.168.0.129:5060 at 14/12/2004 18:21:29:500 (593 bytes): REGISTER sip:192.168.0.129 SIP/2.0 Via: SIP/2.0/UDP 192.168.102.70:5060;branch=z9hG4bK-wg4ok3zkt573;rport From: "snom_01" <sip:snom_01@192.168.0.129>;tag=i7u8p4i1vi To: "snom_01" <sip:snom_01@192.168.0.129> Call-ID: 3c267319f1b3-igmsa9072v8z@192-168-102-70 CSeq: 45683 REGISTER Max-Forwards: 70 Contact: <sip:snom_01@192.168.102.70:5060;line=v8ppcao5&...

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+ /

...horized packets. The password is correct, it connects fine inside the lan but the same username and password fails outside the LAN. <------------> [Nov 24 14:18:41] <--- Transmitting (NAT) to 218.101.6.157:5060 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.50.183:5060;branch=z9hG4bK-26ca393d;received=218.101.6.157 From: 372 <sip:372 at 203.109.148.108>;tag=e25fccc07a79cd65o0 To: 372 <sip:372 at 203.109.148.108>;tag=as1f31845b Call-ID: f4e6d9bc-59a7c482 at 192.168.50.183 CSeq: 26779 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, S...

...re Ver. 2.0.1(563f) sip.conf [8445983] type=friend username=8445983 secret=mypassword nat=0 context=toll-access host=dynamic canreinvite=no reinvite=no allow=ulaw ;allow=alaw mailbox=5983 output from sip debug Sip read: REGISTER sip:192.168.0.5 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.20:5060;branch=z9hG4bK-b1938413 From: 8445985 <sip:8445985@192.168.0.5>;tag=c864004bd9b6bbbdo0 To: 8445985 <sip:8445985@192.168.0.5> Call-ID: 76662903-a6afea65@192.168.0.20 CSeq: 1 REGISTER Max-Forwards: 70 Contact: 8445985 <sip:8445985@192.168.0.20:5060>;expires=9999 User-Agent: Sipura/SPA2000-2.0.11(g...

...an ACK, which asterisk seems to ignore because it retransmits the OK message again Then eventually the phone gives up and sends a BYE message. -- James <--- SIP read from yyy.yyy.yyy.yyy:24050 ---> INVITE sip:dddddddddd at xxx.xxx.xxx.xxx SIP/2.0^M Via: SIP/2.0/UDP 10.1.24.145:7388;branch=z9hG4bK-6e730c81^M From: "sss-sss-ssss" ;tag=bdfe4214c494d109o0^M To: ^M Call-ID: c4560330-de7ca29d at 10.1.24.145^M CSeq: 101 INVITE^M Max-Forwards: 70^M Contact: "sss-sss-ssss" ^M Expires: 240^M User-Agent: Linksys/SPA942-6.1.3(a)^M Content-Length: 395^M Allow: ACK, BYE, CANCEL, INFO,...

...>>Thanks for any help you can offer. >> >>Brian >> >>### START SIP TRACE ### >> >>Sent to udp:192.168.0.129:5060 at 14/12/2004 18:21:29:500 (593 bytes): >> >>REGISTER sip:192.168.0.129 SIP/2.0 >>Via: SIP/2.0/UDP 192.168.102.70:5060;branch=z9hG4bK-wg4ok3zkt573;rport >>From: "snom_01" <sip:snom_01@192.168.0.129>;tag=i7u8p4i1vi >>To: "snom_01" <sip:snom_01@192.168.0.129> >>Call-ID: 3c267319f1b3-igmsa9072v8z@192-168-102-70 >>CSeq: 45683 REGISTER >>Max-Forwards: 70 >>Contact: &l...

...oy *********** PAP2-NA LOG *********** Feb 9 09:00:56 192.168.4.205 Feb 9 09:01:11 192.168.4.205 [0:5060]->192.168.3.14:5060 Feb 9 09:01:11 192.168.4.205 [0:5060]->192.168.3.14:5060 Feb 9 09:01:11 192.168.4.205 NOTIFY sip:192.168.3.14 SIP/2.0^M Via: SIP/2.0/UDP 192.168.4.205:5060;branch=z9hG4bK-96e6cfdd^M From: 1221 < sip:dep2_1221 at 192.168.3.14>;tag=f1611009c41fba9fo0^M To: <sip:192.168.3.14>^M Call-ID: 41e4f931-3594a397 at 192.168.4.205^M CSeq: 14 NOTIFY^M Max-Forwards: 70^M Event: keep-alive^M User-Agent: Linksys/PAP2-3.1.22(LS)^M Content-Length: 0^M ^M Feb 9 09:01:11 19...

...g 0 from snom phone > > > <--- Received SIP request (1230 bytes) from UDP:123.231.72.210:33878 > <http://123.231.72.210:33878> ---> > INVITE sip:0 at 54.206.59.252 <mailto:sip%3A0 at 54.206.59.252>;user=phone SIP/2.0 > Via: SIP/2.0/UDP 123.231.72.210:45835;branch=z9hG4bK-bskkkx1t5bas;rport > From: "outburns00-nhvg5vjjn6-2001" > <sip:outburns00-nhvg5vjjn6-2001 at 54.206.59.252 > <mailto:sip%3Aoutburns00-nhvg5vjjn6-2001 at 54.206.59.252>>;tag=1bb809zgaa > To: <sip:0 at 54.206.59.252 <mailto:sip%3A0 at 54.206.59.252>;user=pho...

I am trying to make PJSIP work with my Cisco SPA504G phone. I have no problems making it work with the chan_sip driver. When I configure my phone, it indicates the contact was added -- Added contact 'sip:7001 at 192.168.9.142:5063' to AOR '7001' with expiration of 3600 seconds Phone shows green light for the line. I then attempt to dial extension 1 and Asterisk crashes.

...Full Contact : sip:105@192.168.0.253:5060 And this is the output of sip debug peer 105 when I call to *98 (for voice messages): asterisk*CLI> sip debug peer 105 SIP Debugging Enabled for IP: 200.93.xxx.xb:5060 Sip read: NOTIFY sip:sip.mydomain.net SIP/2.0 Via: SIP/2.0/UDP 192.168.0.253;branch=z9hG4bK-67ea7370 From: Guillermo Salas M <sip:105@sip.mydomain.net>;tag=4f2df183b116b70c To: <sip:sip.mydomain.net> Call-ID: a584ba93-53c0013c@192.168.0.253 CSeq: 4 NOTIFY Max-Forwards: 70 Event: keep-alive User-Agent: Sipura/SPA2000-2.0.2 Content-Length: 0 10 headers, 0 lines Transmitting (n...

...t=no type=peer context=extensions host=abpusa.com ;disallow=all ;allow=ulaw Here is a sip debug: notice the via for 192.168 in the inbound INVITE, then notice the lack of it on the outbound 200 OK. Sip read: INVITE sip:10@dink.abpusa.com:5070 SIP/2.0 v: SIP/2.0/UDP 209.189.239.106:5060;branch=z9hG4bK-6f04a3acfdd3002c645fcb7605027073.1 v: SIP/2.0/UDP 209.189.239.106:5062;branch=z9hG4bK-dc2a770cf56399c4c0eb8ca964813dc9;nat=true v: SIP/2.0/UDP 192.168.5.102:5060;branch=z9hG4bK-d0t7a9g6nme4;rport=5060;received=4.13.144.17 Record-Route: <sip:abpusa.com:5060;maddr=209.189.239.106;lr=1> Record...

Hi, I'm trying to setup snom 710 phone with asterisk 13 with PJSIP. inbound is working fine but i cannot dial out. i don't hear anything on the phone and asterisk CLI also does not show anything. my config is. please advice. [2001] type=endpoint context=out-local disallow=all allow=ulaw allow=alaw transport=system-udp auth=2001

...ten => 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> &g...

...es work and I see all my online contacts in green. But after a few minutes linphone attempts to renew the subscriptions and asterisk is not happy at all: <--- SIP read from UDP:10.27.128.3:5060 ---> SUBSCRIBE sip:jacques at 10.27.128.1:5060 SIP/2.0 Via: SIP/2.0/UDP 10.27.128.3:5060;branch=z9hG4bK.NYP-ux0Zx;rport From: <sip:john at masked.masked.com>;tag=iGH81k5xf To: <sip:jacques at masked.masked.com>;tag=as3c7de68c CSeq: 22 SUBSCRIBE Call-ID: SQOclJgm4O Max-Forwards: 70 Supported: replaces, outbound Event: presence Expires: 600 Accept: application/pidf+xml Contact: <sip:joh...

...zdev.org iD8DBQFF969jR0exH8dhr/YRAgP0AJ94ygGEPYHtGvLS7McUTrRAP1IkCgCgozv6 rfuVGufsb8wQT3Iwl0ipXNg= =hrcE -----END PGP SIGNATURE----- -------------- next part -------------- <-- SIP read from 172.28.20.4:2051: INVITE sip:374@172.28.2.30;user=phone SIP/2.0 Via: SIP/2.0/UDP 172.28.20.4:2051;branch=z9hG4bK-0wdg2nh9bdfn;rport From: "Test User3" <sip:104@172.28.2.30>;tag=48rd3g03e1 To: <sip:374@172.28.2.30;user=phone> Call-ID: 3c30f7fb186a-n2ptt0oiuivb@snom300-0004132520F7 CSeq: 1 INVITE Max-Forwards: 70 Contact: <sip:104@172.28.20.4:2051;line=7p0ffzkf>;flow-id=1 P-Key-Flags...

Maybe I just woke up too early today. I have SJPhone and X-Lite working perfectly but I cannot for the life of me get sipXphone working properly with Asterisk. Its probably something stupid on my part, but does anyone have a quick setup sheet for it? -Kerry -------------- next part -------------- An HTML attachment was scrubbed... URL:

...gor end but still getting nothing. I looked at sip debug (below) but am new to Asterisk and don't really know what I am looking for. Asterisk works fine with XLITE so I know my installation is ok. Sip read: INVITE sip:90800500005@192.168.0.250 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.1:5060;branch=z9hG4bK-Ifn-9746 From: phone1 <sip:phone1@192.168.0.250:5060>;tag=eSJ-4736 To: <sip:90800500005@192.168.0.250> Call-ID: diY-24872@192.168.1.1 CSeq: 1 INVITE Contact: <sip:phone1@192.168.1.1> Max-Forwards: 70 User-Agent: DrayTek UA-1.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE Content-Type:...

...on > Proxy: 192.168.0.1 > PSTN Line -> Subscriber information > Display name: spaphone > User ID: spaphone > Password: abcde 4. SIP debug output on asterisk console: ============= > REGISTER sip:192.168.0.1 SIP/2.0 > Via: SIP/2.0/UDP 127.0.0.1:5060;branch=z9hG4bK-f05c05f9 > From: spaphone <sip:spaphone at 192.168.0.1>;tag=c2f457c6347c4f23o1 > To: spaphone <sip:spaphone at 192.168.0.1> > Call-ID: f264209-bccc3039 at 127.0.0.1 > CSeq: 38885 REGISTER > Max-Forwards: 70 > Contact: spaphone <sip:spaphone at 127.0.0.1:5060>;exp...