[asterisk-dev] SIP trunking

[Andre Barbosa]

I have to do the same thing to get it work with Huawei Softx3000

Barbosa

John Martin wrote:
> Anton,
>   We've been using Asterisk to trunk to a Huawei switch for over 12
> months now, you may have a later build of the Huawei code but here's
> what we found. 
>
>   You can certainly get odd behaviour - denied was quite a common
> response I seem to remember. We found it doesn't like a number of things
> being sent to it:
> - SIP Fast Update Requests using XML, it needs to see them as old
> fashioned RTCP/H.261
> - silence suppression in SDP, you have to remove:
>
> 	if(!p->owner || !ast_internal_timing_enabled(p->owner))
> 		ast_build_string(&a_audio_next, &a_audio_left,
> "a=silenceSupp:off - - - -\r\n");
>
> from chan_sip.c
>
> Not sure it'll help you, but it worked for us when hooking up to a
> Huawei switch.
>
> There were other valid SIP messages that the Huawei didn't like at the
> time, but nothing that Asterisk used in our installation.
>
> Olle, I think we had a talk about this in Paris and we decided it had to
> stay in the code, but for Huawei it made sense for us. I should also
> probably be on the users forum by now :-)
>
> John
>
>   
>> -----Original Message-----
>> From: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-
>> bounces at lists.digium.com] On Behalf Of Anton
>> Sent: 06 March 2007 16:55
>> To: Asterisk Developers Mailing List
>> Subject: Re: [asterisk-dev] SIP trunking
>>
>> Olle,
>>
>> I'll try to find out from the Huawei what is their SIP
>> trunking is. As I currently know they do use SIP-T as
>> transport, but in specs it's also stated that plain SIP is
>> supported too. Current interconnect behaviour is quite
>> funny, when I try to call their phone - it rings, but if
>> someone pick the phone up, my asterisk starting playing me
>> MOH, and remote side insists that I send them "denied".
>>
>> On 6 March 2007 15:35, Olle E Johansson wrote:
>>     
>>> 1 mar 2007 kl. 17.52 skrev Anton:
>>>       
>>>> any plans to support SIP Trunking?
>>>>         
>>> As previously said, it's a very generic question. The
>>> version of SIP trunking I want to support
>>> is the SIP Forum's SIPconnect specification. It's part of
>>> my work with codename pineapple
>>> (http://www.codename-pineapple.org) to implement an
>>> object called "trunk" in Asterisk.
>>>
>>> Having said that, with proper configuration we can set up
>>> proper SIP trunks with most
>>> equipment today. If you have any specific issue or
>>> variant we do not support, let us
>>> know more details about it.
>>>
>>> Best regards,
>>> /Olle
>>>
>>>
>>> ---
>>> Olle E. Johansson * Asterisk Evangelist, developer * VOOP
>>> A/S olle at voop.com
>>>
>>>
>>>
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