[asterisk-users] asterisk registration
Hi all, I think i understand the problem, actually I have two asterisk server. In the extension.conf file on one server I have added exten = 3923903,1,GOTO(s,1,3923903.conf) which reads the corresponding conf file when ever the extension no. through PSTN is called and learns the location of inbound.php which contains the IVR script to be executed. Now what i want is that through this inbound.php , i should be able to call another asterisk server, where I have also configured twinkle as a softphone. The problems: --I am not able to register this softphone on the previous asterisk server as user 2001, though i modified the server's extension and sip file to include the user 2001 under [phones] context. ---cli chan_sip.c:15839 handle_request_register: Registration from 'user1 sip:2...@172.26.48.208 sip%3a2...@172.26.48.208' failed for '172.26.48.62' - No matching peer found shows this error upon registration.. --at my server it shows 3 unmonitored peers, but the previous server doesn't show any peers on sip show peers..though i have added all three users in sip file, and yes reloaded the dial plan. WARNING[9041]: chan_sip.c:2984 create_addr: No such host: 2001 [Jun 10 12:26:46] WARNING[9041]: app_dial.c:1237 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown) is the error when i do not give ip..assuming 2001 to be registered at the server. when i give the ip of my server.. chan_sip.c:20039 handle_request_invite: Call from '' to extension '2001' rejected because extension not found. is the error..call actually lands up on asterisk server but it shows the above error and ofcourse can not be recieved with softphone. Please help me out in this regard. Though above details may be confusing..I have tried to briefly write in case any more explanation needed, please mail me.I am stuck in this so please help. Thanks in advance Nikhil Kumar summer intern:simmortel voice technologies rit2007033 b.tech IT 6th sem IIIT Allahabad cont...@9793905858 email: rit2007...@iiita.ac.in niksingha...@gmail.com http://profile.iiita.ac.in/RIT2007033/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk registration SIP confusion. Can someone explain this?
We have an Asterisk v1.2.16 box registering with an ITSP using SIP. The registration succeeds, and is confirmed with SIP SHOW REGISTER. However, we frequently (every few minutes) see this on our console: REGISTER attempt 1 to [EMAIL PROTECTED] REGISTER attempt 2 to [EMAIL PROTECTED] Any ideas what is going on? In particular 1. What causes the two register attempt messages above? 2. Why is our asterisk box being associated with the entryunauthorized context, not the entryinternal context? (See below) 3. Why is the contact sip:[EMAIL PROTECTED]:5060 in our SIP messages, why s@ anything? Thanks MD -- Contents of sip.conf at ITSP: [999] context=entryinternal ; I know this context exists! This is the right context. type=friend username=999 secret= callerid=Test 999 host=dynamic nat=no canreinvite=no allow=ulaw allow=alaw dtmfmode=rfc2833 --- Console log from ITSP show strange SIP traffic: --- Scheduling destruction of call mailto:'[EMAIL PROTECTED]' '[EMAIL PROTECTED]' in 15000 ms pbx*CLI pbx*CLI -- SIP read from 123.183.86.231:5060: REGISTER sip:pbx.itsp.com SIP/2.0 Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;rport From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED] Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=999, realm=pbx.itsp.com, algorithm=MD5, uri=sip:pbx.itsp.com, nonce=5cec66c0, response=6451967016fc38f896efeb7247523fe1, opaque= Expires: 120 Contact: sip:[EMAIL PROTECTED]:5060 Event: registration Content-Length: 0 --- (13 headers 0 lines) --- Using latest REGISTER request as basis request Sending to 123.183.86.231 : 5060 (NAT) Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;received=123.183.86.231;rport=506 0 From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED] Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK1a8cee82;received=123.183.86.231;rport=506 0 From: sip:[EMAIL PROTECTED];tag=as3218ff14 To: sip:[EMAIL PROTECTED];tag=as7d680d48 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 103 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 120 Contact: sip:[EMAIL PROTECTED]:5060;expires=120 Date: Fri, 04 May 2007 19:27:58 GMT ontent-Length: 0 -- SIP read from 123.183.86.231:5060: OPTIONS sip:pbx.itsp.com SIP/2.0 Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK36c1df86;rport From: asterisk sip:[EMAIL PROTECTED];tag=as6e5334cf To: sip:pbx.itsp.com Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Fri, 04 May 2007 19:38:36 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- (12 headers 0 lines) --- Looking for s in entryunauthorized (domain pbx.itsp.com) Transmitting (no NAT) to 123.183.86.231:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 123.183.86.231:5060;branch=z9hG4bK36c1df86;received=123.183.86.231;rport=506 0 From: asterisk sip:[EMAIL PROTECTED];tag=as6e5334cf To: sip:pbx.itsp.com;tag=as51d476cd Call-ID: mailto:[EMAIL PROTECTED] [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:74.110.57.25 Accept: application/sdp Content-Length: 0 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk registration
Hi all, some body told me that you can make asterisk to register itself to another asterisk server. i just want to know whether it really can be done or not. i have googled a lot but no answeres. -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk registration
You can by using a register entry in either SIP or IAX connections. Try searching voip-info.org for the details of sip.conf and iax.conf and also connectiong 2 asterisk servers. On 1/17/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, some body told me that you can make asterisk to register itself to another asterisk server. i just want to know whether it really can be done or not. i have googled a lot but no answeres. -- Regards Rizwan Hisham Software Engineer ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Bruce Nortex Networks ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk registration
On 1/18/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, some body told me that you can make asterisk to register itself to another asterisk server. i just want to know whether it really can be done or not. i have googled a lot but no answeres. -- Regards Rizwan Hisham Software Engineer For servers A and B, You need to create a user a/c in say Svr A like rizwan with pwd 1234 and then in svr B sip.conf, put in a line register = rizwan:[EMAIL PROTECTED] You can now create a trunk that uses this a/c to SvrB to terminate calls there See http://www.asterisk.org/doxygen/1.4/Config_sip.html for more info HTH ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Registration as Client to OpenSER
Dear List, http://pastebin.ca/22701This is my problem. Thanks, Ozan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Registration as Client to OpenSER
what is the REGISTER messages are ser saying Iqbal Ozan Blotter wrote: Dear List, http://pastebin.ca/22701 This is my problem. Thanks, Ozan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Registration as Client to OpenSER
Ozan, Put the following to sip_custom.conf: [OpenSER] type=peer username=8333688231 secret=test host=212.154.104.198 fromuser=8333688231; some of the following may not be necessary fromdomain=212.154.104.198 nat=yes dtmfmode=rfc2833 disallow=all allow=g729; whichever codec you want context=from-internal ; this is important, change it, so that outside callers are directed to this context Put the following to sip.conf under general: register = [EMAIL PROTECTED] Put the following to somewhere in from-internal-custom in extensions_custom.conf: exten = _0.,1,Dial(SIP/[EMAIL PROTECTED],60,T) I did all of the above, and registered to YOUR server, and even called 05353490056, but your server gives out: == Everyone is busy/congested at this time (1:0/0/1) probably because there is not enough credit on this test account. This same settings work on my systems. Obviously you are using [EMAIL PROTECTED] Hope this helps, Soner - Original Message - From: Ozan Blotter [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, September 12, 2005 2:46 PM Subject: [Asterisk-Users] Asterisk Registration as Client to OpenSER Dear List, http://pastebin.ca/22701 This is my problem. Thanks, Ozan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Registration as Client to OpenSER
Sorry, I meant: exten = _0.,1,Dial(SIP/[EMAIL PROTECTED],60,T) - Original Message - From: Soner Tari [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, September 12, 2005 3:25 PM Subject: Re: [Asterisk-Users] Asterisk Registration as Client to OpenSER Ozan, Put the following to sip_custom.conf: [OpenSER] type=peer username=8333688231 secret=test host=212.154.104.198 fromuser=8333688231; some of the following may not be necessary fromdomain=212.154.104.198 nat=yes dtmfmode=rfc2833 disallow=all allow=g729; whichever codec you want context=from-internal ; this is important, change it, so that outside callers are directed to this context Put the following to sip.conf under general: register = [EMAIL PROTECTED] Put the following to somewhere in from-internal-custom in extensions_custom.conf: exten = _0.,1,Dial(SIP/[EMAIL PROTECTED],60,T) I did all of the above, and registered to YOUR server, and even called 05353490056, but your server gives out: == Everyone is busy/congested at this time (1:0/0/1) probably because there is not enough credit on this test account. This same settings work on my systems. Obviously you are using [EMAIL PROTECTED] Hope this helps, Soner - Original Message - From: Ozan Blotter [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, September 12, 2005 2:46 PM Subject: [Asterisk-Users] Asterisk Registration as Client to OpenSER Dear List, http://pastebin.ca/22701 This is my problem. Thanks, Ozan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
