[Asterisk-Users] Problem setting SIP incoming/outgoing
Hi I am a newbie to * and I am having a problem which appears strange as I did not find any mention of it anywhere in my search. Simply speaking, I have an external SIP proxy server which I am trying to configure for incoming and outgoing calls from my asterisk installation. So here is my configuration in sip.conf [general] register = user:secret:[EMAIL PROTECTED]:8080 as long as I have just the above entry, I am able to receive incoming calls. Now I would like to setup outgoing calls too. So I create a new section in sip.conf [sipserverout] type=peer secret=secret username=user fromuser=user fromdomain=sipserver.com host=sipserver.com port=8080 context=default with the above configuration I can successfully dial out using dial(SIP/[EMAIL PROTECTED]) but now when I call my incoming number, I get a busy or invalid number signal. If I coment out sipserverout section, I could receive incoming calls again. So I turned on sip debug on CLI. and it appears to me that the following is happening. astreisk takes the incoming call and tries to match it with a section with the same hostname. Now the reverse IP lookup on 109.147.41.48 return sipserver.com (which is correct), so it is trying to send the call to sipserverout which is essentially back to the same server where it came from (Notice the statement Found peer 'sipserverout' in the sip debug logs below). This creates an endless loop and the equipment at the other end terminates the call. According to all the examples I have seen, my setup is the correct setup and everyone seems to be using it. but it does not work for me. I am deperately looking for a solution. Please help. I am using asterisk 1.2.0 beta 1 on FC1. Here is the sip debug dump when a call is coming. -- SIP read from 109.147.41.48:8080: INVITE sip:[EMAIL PROTECTED]:5050 SIP/2.0 Record-Route: sip:209.47.41.48:80;ftag=2C996308-10F9;lr=on Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bK03a4.da6a926.0 Via: SIP/2.0/UDP 209.47.41.61:5060;rport=53084;x-route-tag=tgrp:sroutetor1;branch=z9hG4bK4BB6EA6 From: sip:[EMAIL PROTECTED];tag=2C996308-10F9 To: sip:[EMAIL PROTECTED] Date: Thu, 06 Oct 2005 08:13:58 GMT Call-ID: [EMAIL PROTECTED] Supported: timer Min-SE: 1800 Cisco-Guid: 4208765565-896995802-2793406481-2459445924 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 4 Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Timestamp: 1128586438 Contact: sip:[EMAIL PROTECTED]:53084 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 369 hint: NAThelper hint: SDP rewritten hint: usrloc applied hint: NAT... v=0 o=CiscoSystemsSIP-GW-UserAgent 5168 3221 IN IP4 209.47.41.61 s=SIP Call c=IN IP4 109.147.41.48 t=0 0 m=audio 53870 RTP/AVP 0 8 18 3 101 c=IN IP4 109.147.41.48 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes --- (26 headers 16 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 109.147.41.48 : 80 (non-NAT) Found peer 'sipserverout' Reliably Transmitting (no NAT) to 209.47.41.48:80: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bK03a4.da6a926.0 Via: SIP/2.0/UDP 209.47.41.61:5060;x-route-tag=tgrp:sroutetor1;branch=z9hG4bK4BB6EA6 From: sip:[EMAIL PROTECTED] ;tag=2C996308-10F9 To: sip:[EMAIL PROTECTED] ;tag=as1b7fff99 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED]:5050 Proxy-Authenticate: Digest realm=asterisk, nonce=6d00a83d Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms -- SIP read from 109.147.41.48:8080: ACK sip:[EMAIL PROTECTED]:5050 SIP/2.0 Via: SIP/2.0/UDP 109.147.41.48:8080;branch=z9hG4bK03a4.da6a926.0 From: sip:[EMAIL PROTECTED];tag=2C996308-10F9 Call-ID: [EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as1b7fff99 CSeq: 101 ACK User-Agent: Phone Server 1 Content-Length: 0 ___ --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
[Asterisk-Users] Problem setting SIP incoming/outgoing
Hi I am a newbie to * and I am having a problem which appears strange as I did not find any mention of it anywhere in my search. Simply speaking, I have an external SIP proxy server which I am trying to configure for incoming and outgoing calls from my asterisk installation. So here is my configuration in sip.conf [general] register = user:secret:[EMAIL PROTECTED]:8080 as long as I have just the above entry, I am able to receive incoming calls. Now I would like to setup outgoing calls too. So I create a new section in sip.conf [sipserverout] type=peer secret=secret username=user fromuser=user fromdomain=sipserver.com host=sipserver.com port=8080 context=default with the above configuration I can successfully dial out using dial(SIP/[EMAIL PROTECTED]) but now when I call my incoming number, I get a busy or invalid number signal. If I coment out sipserverout section, I could receive incoming calls again. So I turned on sip debug on CLI. and it appears to me that the following is happening. astreisk takes the incoming call and tries to match it with a section with the same hostname. Now the reverse IP lookup on 109.147.41.48 return sipserver.com (which is correct), so it is trying to send the call to sipserverout which is essentially back to the same server where it came from (Notice the statement Found peer 'sipserverout' in the sip debug logs below). This creates an endless loop and the equipment at the other end terminates the call. According to all the examples I have seen, my setup is the correct setup and everyone seems to be using it. but it does not work for me. I am deperately looking for a solution. Please help. I am using asterisk 1.2.0 beta 1 on FC1. Here is the sip debug dump when a call is coming. -- SIP read from 109.147.41.48:8080: INVITE sip:[EMAIL PROTECTED]:5050 SIP/2.0 Record-Route: sip:209.47.41.48:80;ftag=2C996308-10F9;lr=on Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bK03a4.da6a926.0 Via: SIP/2.0/UDP 209.47.41.61:5060;rport=53084;x-route-tag=tgrp:sroutetor1;branch=z9hG4bK4B B6EA6 From: sip:[EMAIL PROTECTED];tag=2C996308-10F9 To: sip:[EMAIL PROTECTED] Date: Thu, 06 Oct 2005 08:13:58 GMT Call-ID: [EMAIL PROTECTED] Supported: timer Min-SE: 1800 Cisco-Guid: 4208765565-896995802-2793406481-2459445924 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER CSeq: 101 INVITE Max-Forwards: 4 Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Timestamp: 1128586438 Contact: sip:[EMAIL PROTECTED]:53084 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 369 hint: NAThelper hint: SDP rewritten hint: usrloc applied hint: NAT... v=0 o=CiscoSystemsSIP-GW-UserAgent 5168 3221 IN IP4 209.47.41.61 s=SIP Call c=IN IP4 109.147.41.48 t=0 0 m=audio 53870 RTP/AVP 0 8 18 3 101 c=IN IP4 109.147.41.48 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=direction:passive a=nortpproxy:yes --- (26 headers 16 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 109.147.41.48 : 80 (non-NAT) Found peer 'sipserverout' Reliably Transmitting (no NAT) to 209.47.41.48:80: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 209.47.41.48:80;branch=z9hG4bK03a4.da6a926.0 Via: SIP/2.0/UDP 209.47.41.61:5060;x-route-tag=tgrp:sroutetor1;branch=z9hG4bK4BB6EA6 From: sip:[EMAIL PROTECTED] ;tag=2C996308-10F9 To: sip:[EMAIL PROTECTED] ;tag=as1b7fff99 Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED]:5050 Proxy-Authenticate: Digest realm=asterisk, nonce=6d00a83d Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms -- SIP read from 109.147.41.48:8080: ACK sip:[EMAIL PROTECTED]:5050 SIP/2.0 Via: SIP/2.0/UDP 109.147.41.48:8080;branch=z9hG4bK03a4.da6a926.0 From: sip:[EMAIL PROTECTED];tag=2C996308-10F9 Call-ID: [EMAIL PROTECTED] To: sip:[EMAIL PROTECTED];tag=as1b7fff99 CSeq: 101 ACK User-Agent: Phone Server 1 Content-Length: 0 ___ --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] Problem setting SIP incoming/outgoing
I am a newbie to * and I am having a problem which appears strange as I did
not find any mention of it anywhere in my search.
Simply speaking, I have an external SIP proxy server which I am trying to
configure for incoming and outgoing calls from my asterisk installation. So
here is my configuration in sip.conf
[general]
register = user:secret:[EMAIL PROTECTED]:8080
as long as I have just the above entry, I am able to receive incoming calls.
Now I would like to setup outgoing calls too. So I create a new section in
sip.conf
[sipserverout]
type=peer
secret=secret
username=user
fromuser=user
fromdomain=sipserver.com
host=sipserver.com
port=8080
context=default
with the above configuration I can successfully dial out using
dial(SIP/[EMAIL PROTECTED])
but now when I call my incoming number, I get a busy or invalid number
signal. If I coment out sipserverout section, I could receive incoming calls
again.
So I turned on sip debug on CLI. and it appears to me that the following is
happening. astreisk takes the incoming call and tries to match it with a
section with the same hostname. Now the reverse IP lookup on 109.147.41.48
return sipserver.com (which is correct), so it is trying to send the call to
sipserverout which is essentially back to the same server where it came from
(Notice the statement Found peer 'sipserverout' in the sip debug logs
below). This creates an endless loop and the equipment at the other end
terminates the call.
According to all the examples I have seen, my setup is the correct setup and
everyone seems to be using it. but it does not work for me. I am deperately
looking for a solution. Please help.
I am using asterisk 1.2.0 beta 1 on FC1.
In very general terms, you probably want something like this in your sip.conf:
[sipserver]
type=friend
secret=secret
username=user
fromuser=user
fromdomain=sipserver.com
host=sipserver.com
port=8080
insecure=very
canreinvite=no
dtmfmode=inband
context=from-sipserver
disallow=all
allow=ulaw
For sip stuff, notice the use of type=friend and canreinvite=no. The use
of the register statement (in this case) implies use of type=friend (for
both incoming and outgoing calls).
Then in extensions.conf, use something like this:
exten = _1NX,3,Dial(SIP/sipserver/${EXTEN})
where SIP/sipserver is referring to the context [sipserver] in sip.conf.
Did the folks at sipserver.com tell you to use port=8080? If not,
remove that statement as the default for sip is port=5060.
There are other ways to accomplish the same thing, so consider the above
as only way to do it.
___
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