Asterisk Transfer()

[Kostya]

Transfer() is causing Asterisk to crash on both the 1.2 and 1.4 versions. Here is my dialplan:

When a call comes in it enters the _X. extension and dials a SIP system at 10.10.1.95. The SIP systems then refers the call to 18529999999999, at which point the call enters the _1852XXXXXXXXXX extension. The transfer completes, but crashes the Asterisk system every 2-3 calls. The extenion plan and warnings I see are printed below.

exten => _1852XXXXXXXXXX,1,Answer()
exten => _1852XXXXXXXXXX,n,Transfer(SIP/${EXTEN}@PROVIDER)
exten => _1852XXXXXXXXXX,n,Hangup()

exten => _X.,1,Answer()
exten => _X.,n,Dial(SIP/transferTestPROVIDER@10.10.1.95)
exten => _X.,n,Hangup()

[Dec 10 13:46:54] WARNING[5072]: chan_sip.c:11997 handle_response: Remote host can’t match request NOTIFY to call ‘4155de7e11251806114397537a39791a@10.10.1.71’. Giving up.
[Dec 10 13:46:54] WARNING[5072]: chan_sip.c:11997 handle_response: Remote host can’t match request NOTIFY to call ‘4155de7e11251806114397537a39791a@10.10.1.71’. Giving up.
[Dec 10 13:47:05] WARNING[5072]: chan_sip.c:11991 handle_response: Remote host can’t match request BYE to call ‘1894f78e3575d545405ef7673002e80c@10.10.1.71’. Giving up.

Thanks for any advice.

[dmikusa]

Perhaps consider using the Dial command instead of using Transfer.

There are very few circumstances where you would want to use Transfer instead of Dial. I don’t know your situation so I can’t say for sure, but you may want to give it a try and see if it fixes the problem.

Dan

[Kostya]

I need to do a SIP REFER and release the call from Asterisk.

I think the problem has to do with the incoming SIP refer. Asterisk starts by dialing a system, and that systems does its own Transfer() which Asterisk may not interpret well.

[dmikusa]

I had attempted to do a similar project to yours a while back, and had tons of problems with it. I wanted one Asterisk box to answer the call and intelligently transfer it to another box, removing itself from the loop for efficiency.

It because too much of a pain to get the machine out of the loop, so I just compromised and left the asterisk box in the loop and turned on “canreinvite” in sip.conf, which routes the media (SDP) directly between endpoints.

I have heard of some people using another program called Sip Express Router (SER) to do what you are describing, but I don’t really know much more about it.

http://www.voip-info.org/wiki/index.php?page=SIP+Express+Router

If you can think of a nicer solution, I’d be interested to hear it.

Dan

[Kostya]

Is SER similar to SipX?

[dmikusa]

The sipX solution is an Enterprise SIP Open Source PBX VoIP solution complete with voice mail and auto-attendant. Alternatively, sipX can be used as a high performance Enterprise toll-bypass SIP router. It combines all common calling features, XML-based SIP call routing, voice mail and auto-attendant, Web-based configuration, as well as integrated management and configuration of the PBX and attached phones and gateways. sipX is a modular server based solution that runs on standard Linux. sipX does not require any additional hardware as it interoperates with any SIP compliant gateway, phone or application. sipX is a native SIP communications solution strictly following and implementing all the relevant SIP IETF standards.

[quote]SIP Express Router (ser) is a high-performance, configurable, free SIP ( RFC3261 ) server . It can act as registrar, proxy or redirect server. SER features an application-server interface, presence support, SMS gateway, SIMPLE2Jabber gateway, RADIUS/syslog accounting and authorization, server status monitoring, FCP security, etc. Web-based user provisioning, serweb, available.

Its performance allows it to deal with operational burdens, such as broken network components, attacks, power-up reboots and rapidly growing user population.

SER’s configuration ability meets needs of a whole range of scenarios including small-office use, enterprise PBX replacements and carrier services. [/quote]

I believe that SER is all written in C and is reported to be very fast and very efficient. While SipX is written with a mix of C++ and Java and has more demanding hardware requirements.

Dan