search for: 1103m

...talled. Has anyone successfully configured Asterisk on OS X with FreeWorld Dialup? Do I need to use an outbound proxy? If so, how do I configure that? Any help or suggestions are greatly appreciated. Thank in advance. Kenton Log: ? 2004 Xten Networks, Inc. All rights reserved. X-Lite release 1103m build stamp 14266 License key: A91031380B8611D99AAB000393BE4F08 Established SIP protocol listen on: 172.30.247.226:5060 Discovered Symmetric NAT Firewall SIP: 172.30.247.226:5060 RTP: 172.30.247.226:8000 NAT: 63.211.54.166 PROXY#0: 172.30.247.226:5060 SEND TIME: 36063869 SEND >> 172.30....

...om: timtest <sip:1111@192.168.8.151>;tag=109208562 To: timtest <sip:1111@192.168.8.151> Contact: "timtest" <sip:1111@192.168.8.203:5060> Call-ID: C99F00F557AE457D85513CC954CE72D3@192.168.8.151 CSeq: 22204 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 SEND TIME: 341552005 SEND >> 192.168.8.151:5060 REGISTER sip:192.168.8.151 SIP/2.0 Via: SIP/2.0/UDP 192.168.8.203:5060;rport;branch=z9hG4bK3FB2D9058F8B41F79E68D710E384C12A From: timtest <sip:1111@192.168.8.151>;tag=109208562 To: timtest <sip:1111@192.168.8.151>...

...to my address. There is not a firewall on the PBX yet. here is the log off the softphone..and the PBX logs look like this Registration from 'Tommy Denton <sip:201@209.101.93.30>' failed for ' 24.0.114.xxx' (c) 2004 Xten Networks, Inc. All rights reserved. X-Lite release 1103m build stamp 14262 License key: 258C984DF72244D39564431814E958A1 Established SIP protocol listen on: 192.168.10.8:5060<http://192.168.10.8:5060> Discovered Port Restricted Cone NAT Firewall SIP: 192.168.10.8:5060 <http://192.168.10.8:5060> RTP: 192.168.10.8:8000 <http://192.168.10....

...848F9808AD84E829CA819 From: rdelite <sip:200@10.0.0.201>;tag=3097086592 To: <sip:1234@10.0.0.201> Contact: <sip:200@10.0.0.250:5060> Call-ID: A26B9D85-1A5C-49DD-8507-B15B041B6C35@10.0.0.250 CSeq: 6629 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 242 v=0 o=200 15532664 15532804 IN IP4 10.0.0.250 s=X-Lite c=IN IP4 10.0.0.250 t=0 0 m=audio 8000 RTP/AVP 3 97 110 101 a=rtpmap:3 gsm/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:110 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 RECEIVE TIME: 15532855 RECEIVE << 10.0...

...:damencho@194.12.230.167>;tag=4066431665 To: damencho <sip:damencho@194.12.230.167> Contact: "damencho" <sip:damencho@213.240.242.42:5060> Call-ID: 7FEA34DBA7E1495F94AEC70F236290EC@194.12.230.167 CSeq: 58756 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 11 headers, 0 lines Using latest request as basis request Sending to 213.240.242.42 : 5060 (non-NAT) Transmitting (NAT): SIP/2.0 100 Trying Via: SIP/2.0/UDP 213.240.242.42:5060;rport;branch=z9hG4bKBDF5CC31592C4472A643BDD8C314BD09;received=213.240.242.42 From: damencho <sip:d...

Hello, I have a one-way audio problem. If any one can give me a clue on how to solve it, I'd highly appreciate. My configuration is: Both Asterisk server and a SIP phone run within a LAN. Asterisk: CVS-HEAD-06/27/04-11:42:23. SIP phone is X-Lite release 1103m build stamp 14262. The Linux box that running Asterisk server is RedHat 2.4.18-14. Asterisk server runs on IP: 192.168.1.102. X-Lite (phone A) is on Win2K, with IP 192.168.1.100. They are both behind a router with dynamic IP address. Assume its public IP is aaa.bbb.ccc.ddd. I have another X_Lite...

...: "asterisk" <sip:asterisk@192.168.x.x>;tag=as6a4689e3 To: <sip:192.168.2.50>;tag=1713780919 Contact: <sip:xlite1@192.168.2.50:5060> Call-ID: 2edd9eef1e40bad20f48302e4a1d673a@192.168.x.x Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,NOTIFY CSeq: 102 OPTIONS Server: X-Lite release 1103m Content-Length: 0 Any reasons why I can't place a call. Thanks, Geoff

Hi, I have the following situation: I've installed Asterisk at Machine 1 (M1 - IP: 192.168.1.145) and X-Lite (X_lite-Xten-Win32-1103m.exe from www.xten.com) at Machine 2 (M2 - IP: 192.168.1.100) and Machine 3 (M3 - IP: 192.168.1.200). I need to catch the SIP and MGCP messages that will appear when M2 calls to M3 and vice versa. The SIP messages are working (I don't have problems with the config file), but I found some problem...

...s the IP Address of the new install. I can ping that box. When I try to connect I get hung on the "Awaiting Proxy login information" and the log reads: ======================================================================== ? 2004 Xten Networks, Inc. All rights reserved. X-Lite release 1103m build stamp 14262 License key: A27D1192D9FA4B609F02F3AC31B6BD12 Established SIP protocol listen on: 172.16.17.99:5060 Discovered Port Restricted Single Mapped Port Symmetric NAT Firewall SIP: 172.16.17.99:5060 RTP: 172.16.17.99:8000 NAT: 204.94.248.12 Discovering external SIP port on symmetric...

...e error that I had previously had had to a configuration problem. Start asterisk in modality consol and when two softphone speaks is not felt well, and I have the following error: -- Registered SIP '1000' at 10.0.0.7 port 5060 expires 1800 -- Saved useragent "X-Lite release 1103m" for peer 1000 -- Registered SIP '1001' at 10.0.0.5 port 5060 expires 1800 -- Saved useragent "X-Lite release 1103m" for peer 1001 -- Executing Dial("SIP/1001-73df", "sip/1000|20|rt") in new stack -- Called 1000 -- SIP/1000-60e3 is ring...

...g 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 host = sip.net2phone.com username = PHONENUMBER secret = PASSWORD fromuser = PHONENUMBER fromdomain = net2phone.com context = incoming insecure = very canreinvite = no *extensions.conf* [outgoing] exten => _9NXXNXXXX...

...how can the caller (using mu-law) hear the voicemail prompts? Would Asterisk be doing a half duplex protocol conversion? Any insight would be greatly appreciated!! Current configuration: Fedora Core 1 Asterisk - 1.0.7 (had same problem on 1.0.6) SJPhone - 1.50.271d, Mar 11 2005 (WinXP) XLite - 1103m build stamp 14262 (WinXP) Zultys Zip2 - ZUTS 3.52 sip.conf exerpt: [6003] ; (A) type=friend regexten=6003 username=6003 host=dynamic disallow=all ;allow=gsm allow=ulaw [6004] ; (C) type=friend regexten=6004 username=6004 host=dynamic disallow=all ;allow=gsm allow=ulaw [2101] ; (B) type=...

Thank you Melissa. I love the phone but the dial keypad is a little bouncy. I was hoping for a more solid feel like on the analog PT390's or my quality standard, the Nortel 9417CW. Other than the MWI problem, I'd like more documentation on the configuration paramters. I have found little online configuration documentation other than very basic stuff on the Sayson website. I'd

...ister with asterisk and you can see the message on the asterisk console. The command sip show users will show you a list of registered users Asterisk Ready. *CLI> -- Registered SIP 'jackie.clough' at 192.168.1.13 port 5060 expires 180 -- Saved useragent "X-Lite release 1103m" for peer jackie.clough *CLI> sip show users Username Secret Accountcode Def.Context ACL NAT rons.desk.at.in xxxxxxxx from_SIP_extens No RFC35 ians.desk.at.in xxxxxxxx from_SIP_extens No RFC35 jackie.clough...

...lt;sip:100@public.asterisk.ip.addr>;tag=as353ed737 To: <sip:143@priv.client.ip.addr:5060>;tag=3772325084 Contact: <sip:143@priv.client.ip.addr:5060> Call-ID: 34b745fa12f29b542360765253aaa037@public.asterisk.ip.addr CSeq: 102 INVITE Content-Type: application/sdp Server: X-Lite release 1103m Content-Length: 294 v=0 o=143 8264473 8266947 IN IP4 public.client.ip.addr s=X-Lite c=IN IP4 public.client.ip.addr t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000...

...nf: context=frompstnisdn This works ok on another asterisk box I setup. But on incoming calls I get: -- Extension '787367' in context 'frompstnisdn' from '07768385144' does not exist. Rejecting call on channel 0/1, span 1 -- Saved useragent "X-Lite release 1103m" for peer 202 -- Extension '787367' in context 'frompstnisdn' from '07768385144' does not exist. Rejecting call on channel 0/1, span 1 Do I need to enable something to be able to use the s in extensions.conf? Angus

...From: Tester <sip:1000@10.100.249.12>;tag=3354744682 To: Tester <sip:1000@10.100.249.12> Contact: "Tester" <sip:1000@10.100.249.86:5060> Call-ID: 6A1715C235994196A7739A624B6D0C41@10.100.249.12 CSeq: 4806 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 RECEIVE TIME: 10157385 RECEIVE << 10.100.249.12:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.100.249.86:5060;branch=z9hG4bKFAC1B6F2B5414EE9855696A09A83FB22 From: Tester <sip:1000@10.100.249.12>;tag=3354744682 To: Tester <sip:1000@10.100.249.12>;tag=as7ae925e2...

...62A83A419FF9A75 From: 111 <sip:111@10.1.0.254>;tag=3904243491 To: 111 <sip:111@10.1.0.254> Contact: "111" <sip:111@10.1.0.111:5060> Call-ID: 22493486CC3B43E98157502206FFD604@10.1.0.254 CSeq: 43039 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1103m Content-Length: 0 RECEIVE TIME: 17729573 RECEIVE << 10.1.0.254:5060 SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.1.0.111:5060;branch=z9hG4bK1E46C36B17664F1BA62A83A419FF9A75 From: 111 <sip:111@10.1.0.254>;tag=3904243491 To: 111 <sip:111@10.1.0.254>;tag=as69260c6b Call-ID: 2...

...0460D92CD From: 1 <sip:Phone1@192.168.2.120>;tag=570805602 To: <sip:2123@192.168.2.120> Contact: <sip:Phone1@192.168.2.103:5060> Call-ID: 5C01A7C0-1D67-11DA-9217-0800460D92CD@192.168.2.103 CSeq: 24637 INVITE Max-Forwards: 70 Content-Type: application/sdp User-Agent: X-Lite release 1103m Content-Length: 297 v=0 o=Phone1 22215362 22215384 IN IP4 192.168.2.103 s=X-Lite c=IN IP4 192.168.2.103 t=0 0 m=audio 8000 RTP/AVP 0 8 3 98 97 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:3 gsm/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:97 speex/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101...

Hi Is there a phone comparison matrix I could consult I have a series of features that I would like to evaluate on the most common phones on the market example: dual-ethernet POE / direct power / both number of lines speed dials programmable buttons BLF LEDS Headset plug conference call built in hands free operation display size codecs communication protocol (SIP, h.323) price availability