External transfer fails (from Asterisk)

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External transfer fails (from Asterisk)

Peter den Hartog
Hello,

I don't know if i'm on the right mailing list for this issue but maby i'm not the only one that had it :-).

I implemented opensips and it works good, the normal calls are going great, outside/inside it all works. inside transfer (exten to exten) works to.

But when an outside caller calls the office, it goes to the asterisk, and asterisk forwards it to an opensips extension. exten = x,Dial,1,(SIP/202@opensips.org) That works great, the caller gets the right person, but when the one being called, transfer that call it gone.

I think it's because asterisk is trying to transfer this caller, but the extension is not there (it's in opensips ofcourse, but not in *)

I can connect the asterisk users to the opensips users by connecting the database, but is this really needed? or is there another issue here? Do i miss something?
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Re: External transfer fails (from Asterisk)

Bogdan-Andrei Iancu
Hi Peter,

Peter den Hartog wrote:
> Hello,
>
> I don't know if i'm on the right mailing list for this issue but maby i'm
> not the only one that had it :-).
>  
if it is opensips related, you are on the right list :)
> I implemented opensips and it works good, the normal calls are going great,
> outside/inside it all works. inside transfer (exten to exten) works to.
>
> But when an outside caller calls the office, it goes to the asterisk, and
> asterisk forwards it to an opensips extension. exten =
> x,Dial,1,(SIP/[hidden email]) That works great, the caller gets the right
> person, but when the one being called, transfer that call it gone.
>  
This is the first scenario where * is fronting OpenSIPS ...typically is
the other way around :D
> I think it's because asterisk is trying to transfer this caller, but the
> extension is not there (it's in opensips ofcourse, but not in *)
>  
Normally, the call transfer (from the phone) is done via a REFER request
(inside the ongoing dialog) - What I suspect is that , as * is in the
path of all calls with external users, * will intercept the REFER and
try to handle it locally.

Try to get a trace and see if this is what happens = REFER being
consumed by *, instead of passing it to the external party.

Regards,
Bogdan
> I can connect the asterisk users to the opensips users by connecting the
> database, but is this really needed? or is there another issue here? Do i
> miss something?
>  


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Re: External transfer fails (from Asterisk)

Peter den Hartog

Bogdan-Andrei Iancu wrote
Hi Peter,

Peter den Hartog wrote:
> Hello,
>
> I don't know if i'm on the right mailing list for this issue but maby i'm
> not the only one that had it :-).
>  
if it is opensips related, you are on the right list :)
> I implemented opensips and it works good, the normal calls are going great,
> outside/inside it all works. inside transfer (exten to exten) works to.
>
> But when an outside caller calls the office, it goes to the asterisk, and
> asterisk forwards it to an opensips extension. exten =
> x,Dial,1,(SIP/202@opensips.org) That works great, the caller gets the right
> person, but when the one being called, transfer that call it gone.
>  
This is the first scenario where * is fronting OpenSIPS ...typically is
the other way around :D
> I think it's because asterisk is trying to transfer this caller, but the
> extension is not there (it's in opensips ofcourse, but not in *)
>  
Normally, the call transfer (from the phone) is done via a REFER request
(inside the ongoing dialog) - What I suspect is that , as * is in the
path of all calls with external users, * will intercept the REFER and
try to handle it locally.

Try to get a trace and see if this is what happens = REFER being
consumed by *, instead of passing it to the external party.

Regards,
Bogdan
> I can connect the asterisk users to the opensips users by connecting the
> database, but is this really needed? or is there another issue here? Do i
> miss something?
>  


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Users@lists.opensips.org
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Hello Bogdan,

That is correct,
in Asterisk i see nothing of a new call, or a transfer.. but the phone is creating a new call on line 2, in opensips i just see a new ongoing call. (the line 2 call) and on the outside phone i hear the asterisk wait/hold music.

Is there any smart solution for this? can i just forward the complete call to opensips and let asterisk only forward it, and not create the call? (it now just does a dial to the sip member in opensips)
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Re: External transfer fails (from Asterisk)

Peter den Hartog

Peter den Hartog wrote
Bogdan-Andrei Iancu wrote
Hi Peter,

Peter den Hartog wrote:
> Hello,
>
> I don't know if i'm on the right mailing list for this issue but maby i'm
> not the only one that had it :-).
>  
if it is opensips related, you are on the right list :)
> I implemented opensips and it works good, the normal calls are going great,
> outside/inside it all works. inside transfer (exten to exten) works to.
>
> But when an outside caller calls the office, it goes to the asterisk, and
> asterisk forwards it to an opensips extension. exten =
> x,Dial,1,(SIP/202@opensips.org) That works great, the caller gets the right
> person, but when the one being called, transfer that call it gone.
>  
This is the first scenario where * is fronting OpenSIPS ...typically is
the other way around :D
> I think it's because asterisk is trying to transfer this caller, but the
> extension is not there (it's in opensips ofcourse, but not in *)
>  
Normally, the call transfer (from the phone) is done via a REFER request
(inside the ongoing dialog) - What I suspect is that , as * is in the
path of all calls with external users, * will intercept the REFER and
try to handle it locally.

Try to get a trace and see if this is what happens = REFER being
consumed by *, instead of passing it to the external party.

Regards,
Bogdan
> I can connect the asterisk users to the opensips users by connecting the
> database, but is this really needed? or is there another issue here? Do i
> miss something?
>  


_______________________________________________
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Hello Bogdan,

That is correct,
in Asterisk i see nothing of a new call, or a transfer.. but the phone is creating a new call on line 2, in opensips i just see a new ongoing call. (the line 2 call) and on the outside phone i hear the asterisk wait/hold music.

Is there any smart solution for this? can i just forward the complete call to opensips and let asterisk only forward it, and not create the call? (it now just does a dial to the sip member in opensips)

Oke a little update, i can now do blind (cold) transfers from asterisk to opensips (outside lines) but not hot transfers, then the call gets disconnected.
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Re: External transfer fails (from Asterisk)

Bogdan-Andrei Iancu
In reply to this post by Peter den Hartog
Hi Peter,

Peter den Hartog wrote:

>
> Bogdan-Andrei Iancu wrote:
>  
>> Hi Peter,
>>
>> Peter den Hartog wrote:
>>    
>>> Hello,
>>>
>>> I don't know if i'm on the right mailing list for this issue but maby i'm
>>> not the only one that had it :-).
>>>  
>>>      
>> if it is opensips related, you are on the right list :)
>>    
>>> I implemented opensips and it works good, the normal calls are going
>>> great,
>>> outside/inside it all works. inside transfer (exten to exten) works to.
>>>
>>> But when an outside caller calls the office, it goes to the asterisk, and
>>> asterisk forwards it to an opensips extension. exten =
>>> x,Dial,1,(SIP/[hidden email]) That works great, the caller gets the
>>> right
>>> person, but when the one being called, transfer that call it gone.
>>>  
>>>      
>> This is the first scenario where * is fronting OpenSIPS ...typically is
>> the other way around :D
>>    
>>> I think it's because asterisk is trying to transfer this caller, but the
>>> extension is not there (it's in opensips ofcourse, but not in *)
>>>  
>>>      
>> Normally, the call transfer (from the phone) is done via a REFER request
>> (inside the ongoing dialog) - What I suspect is that , as * is in the
>> path of all calls with external users, * will intercept the REFER and
>> try to handle it locally.
>>
>> Try to get a trace and see if this is what happens = REFER being
>> consumed by *, instead of passing it to the external party.
>>
>> Regards,
>> Bogdan
>>    
>>> I can connect the asterisk users to the opensips users by connecting the
>>> database, but is this really needed? or is there another issue here? Do i
>>> miss something?
>>>  
>>>      
>> _______________________________________________
>> Users mailing list
>> [hidden email]
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>>
>>    
>
> Hello Bogdan,
>
> That is correct,
> in Asterisk i see nothing of a new call, or a transfer.. but the phone is
> creating a new call on line 2, in opensips i just see a new ongoing call.
> (the line 2 call) and on the outside phone i hear the asterisk wait/hold
> music.
>  
when doing call transfer via REFER, the REFER is propagating to the
other party and the other party is responsible foe generating the new
call - but as you have the Asteirsk on the path,it will behave as a end
point, so * must generate the new call.
> Is there any smart solution for this? can i just forward the complete call
> to opensips and let asterisk only forward it, and not create the call? (it
> now just does a dial to the sip member in opensips)
>  
hmmm...not following here..could you detail a bit?

Regards,
Bogdan





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Re: External transfer fails (from Asterisk)

Bogdan-Andrei Iancu
In reply to this post by Peter den Hartog
Peter den Hartog wrote:

>
> Peter den Hartog wrote:
>  
>>
>> Bogdan-Andrei Iancu wrote:
>>    
>>> Hi Peter,
>>>
>>> Peter den Hartog wrote:
>>>      
>>>> Hello,
>>>>
>>>> I don't know if i'm on the right mailing list for this issue but maby
>>>> i'm
>>>> not the only one that had it :-).
>>>>  
>>>>        
>>> if it is opensips related, you are on the right list :)
>>>      
>>>> I implemented opensips and it works good, the normal calls are going
>>>> great,
>>>> outside/inside it all works. inside transfer (exten to exten) works to.
>>>>
>>>> But when an outside caller calls the office, it goes to the asterisk,
>>>> and
>>>> asterisk forwards it to an opensips extension. exten =
>>>> x,Dial,1,(SIP/[hidden email]) That works great, the caller gets the
>>>> right
>>>> person, but when the one being called, transfer that call it gone.
>>>>  
>>>>        
>>> This is the first scenario where * is fronting OpenSIPS ...typically is
>>> the other way around :D
>>>      
>>>> I think it's because asterisk is trying to transfer this caller, but the
>>>> extension is not there (it's in opensips ofcourse, but not in *)
>>>>  
>>>>        
>>> Normally, the call transfer (from the phone) is done via a REFER request
>>> (inside the ongoing dialog) - What I suspect is that , as * is in the
>>> path of all calls with external users, * will intercept the REFER and
>>> try to handle it locally.
>>>
>>> Try to get a trace and see if this is what happens = REFER being
>>> consumed by *, instead of passing it to the external party.
>>>
>>> Regards,
>>> Bogdan
>>>      
>>>> I can connect the asterisk users to the opensips users by connecting the
>>>> database, but is this really needed? or is there another issue here? Do
>>>> i
>>>> miss something?
>>>>  
>>>>        
>>> _______________________________________________
>>> Users mailing list
>>> [hidden email]
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>      
>> Hello Bogdan,
>>
>> That is correct,
>> in Asterisk i see nothing of a new call, or a transfer.. but the phone is
>> creating a new call on line 2, in opensips i just see a new ongoing call.
>> (the line 2 call) and on the outside phone i hear the asterisk wait/hold
>> music.
>>
>> Is there any smart solution for this? can i just forward the complete call
>> to opensips and let asterisk only forward it, and not create the call? (it
>> now just does a dial to the sip member in opensips)
>>
>>    
>
>
> Oke a little update, i can now do blind (cold) transfers from asterisk to
> opensips (outside lines) but not hot transfers, then the call gets
> disconnected.
>  
Do you see some NOTIFY requests going around? they are used during
attended transfer to inform on the new call state.....

Regards,
Bogdan

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[hidden email]
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
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Re: External transfer fails (from Asterisk)

Peter den Hartog

Bogdan-Andrei Iancu wrote
Peter den Hartog wrote:
>
> Peter den Hartog wrote:
>  
>>
>> Bogdan-Andrei Iancu wrote:
>>    
>>> Hi Peter,
>>>
>>> Peter den Hartog wrote:
>>>      
>>>> Hello,
>>>>
>>>> I don't know if i'm on the right mailing list for this issue but maby
>>>> i'm
>>>> not the only one that had it :-).
>>>>  
>>>>        
>>> if it is opensips related, you are on the right list :)
>>>      
>>>> I implemented opensips and it works good, the normal calls are going
>>>> great,
>>>> outside/inside it all works. inside transfer (exten to exten) works to.
>>>>
>>>> But when an outside caller calls the office, it goes to the asterisk,
>>>> and
>>>> asterisk forwards it to an opensips extension. exten =
>>>> x,Dial,1,(SIP/202@opensips.org) That works great, the caller gets the
>>>> right
>>>> person, but when the one being called, transfer that call it gone.
>>>>  
>>>>        
>>> This is the first scenario where * is fronting OpenSIPS ...typically is
>>> the other way around :D
>>>      
>>>> I think it's because asterisk is trying to transfer this caller, but the
>>>> extension is not there (it's in opensips ofcourse, but not in *)
>>>>  
>>>>        
>>> Normally, the call transfer (from the phone) is done via a REFER request
>>> (inside the ongoing dialog) - What I suspect is that , as * is in the
>>> path of all calls with external users, * will intercept the REFER and
>>> try to handle it locally.
>>>
>>> Try to get a trace and see if this is what happens = REFER being
>>> consumed by *, instead of passing it to the external party.
>>>
>>> Regards,
>>> Bogdan
>>>      
>>>> I can connect the asterisk users to the opensips users by connecting the
>>>> database, but is this really needed? or is there another issue here? Do
>>>> i
>>>> miss something?
>>>>  
>>>>        
>>> _______________________________________________
>>> Users mailing list
>>> Users@lists.opensips.org
>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>>      
>> Hello Bogdan,
>>
>> That is correct,
>> in Asterisk i see nothing of a new call, or a transfer.. but the phone is
>> creating a new call on line 2, in opensips i just see a new ongoing call.
>> (the line 2 call) and on the outside phone i hear the asterisk wait/hold
>> music.
>>
>> Is there any smart solution for this? can i just forward the complete call
>> to opensips and let asterisk only forward it, and not create the call? (it
>> now just does a dial to the sip member in opensips)
>>
>>    
>
>
> Oke a little update, i can now do blind (cold) transfers from asterisk to
> opensips (outside lines) but not hot transfers, then the call gets
> disconnected.
>  
Do you see some NOTIFY requests going around? they are used during
attended transfer to inform on the new call state.....

Regards,
Bogdan

_______________________________________________
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Nope, no NOTIFY requests.

Well wat is ment was making asterisk dumb, and just let if forward a complete call.. so instead of doing a dial to an opensips extention, just make a full transfer of the call to the opensips server, and then to the extention.

I'm trying it the other way arround now, as you said earlier that the opensips recieves all the calls (so is directly connected to the sip trunk) but i have some strange issue's with that 2, i can't call outside and when i call inside, the phone rings (i just made a alias) and then i can't pick it up or anything, the phone doesn't respond!

Any ideas ?
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Re: External transfer fails (from Asterisk)

Bogdan-Andrei Iancu
Peter den Hartog wrote:

>>>>>    
>>>>>          
>>>> Hello Bogdan,
>>>>
>>>> That is correct,
>>>> in Asterisk i see nothing of a new call, or a transfer.. but the phone
>>>> is
>>>> creating a new call on line 2, in opensips i just see a new ongoing
>>>> call.
>>>> (the line 2 call) and on the outside phone i hear the asterisk wait/hold
>>>> music.
>>>>
>>>> Is there any smart solution for this? can i just forward the complete
>>>> call
>>>> to opensips and let asterisk only forward it, and not create the call?
>>>> (it
>>>> now just does a dial to the sip member in opensips)
>>>>
>>>>    
>>>>        
>>> Oke a little update, i can now do blind (cold) transfers from asterisk to
>>> opensips (outside lines) but not hot transfers, then the call gets
>>> disconnected.
>>>  
>>>      
>> Do you see some NOTIFY requests going around? they are used during
>> attended transfer to inform on the new call state.....
>>
>>
>>    
> Nope, no NOTIFY requests.
>
> Well wat is ment was making asterisk dumb, and just let if forward a
> complete call.. so instead of doing a dial to an opensips extention, just
> make a full transfer of the call to the opensips server, and then to the
> extention.
>
> I'm trying it the other way arround now, as you said earlier that the
> opensips recieves all the calls (so is directly connected to the sip trunk)
> but i have some strange issue's with that 2, i can't call outside and when i
> call inside, the phone rings (i just made a alias) and then i can't pick it
> up or anything, the phone doesn't respond!
>  
Hi Peter,

have you checked the SIP trace to see why the call is not established
when you pick up the ringing phone ?

Regards,
Bogdan

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Re: External transfer fails (from Asterisk)

Peter den Hartog
Yes i did, but it makes no sence to me. I fixed it tho with giving my opensips a private network + modem, for testing. But i have another issue with that, i can call outside, but the inside gives to many hops, i opened a new message about this: http://n2.nabble.com/incoming-calls-fail-from-outside-td3821099.html#a3821099


Bogdan-Andrei Iancu wrote
Peter den Hartog wrote:
>>>>>    
>>>>>          
>>>> Hello Bogdan,
>>>>
>>>> That is correct,
>>>> in Asterisk i see nothing of a new call, or a transfer.. but the phone
>>>> is
>>>> creating a new call on line 2, in opensips i just see a new ongoing
>>>> call.
>>>> (the line 2 call) and on the outside phone i hear the asterisk wait/hold
>>>> music.
>>>>
>>>> Is there any smart solution for this? can i just forward the complete
>>>> call
>>>> to opensips and let asterisk only forward it, and not create the call?
>>>> (it
>>>> now just does a dial to the sip member in opensips)
>>>>
>>>>    
>>>>        
>>> Oke a little update, i can now do blind (cold) transfers from asterisk to
>>> opensips (outside lines) but not hot transfers, then the call gets
>>> disconnected.
>>>  
>>>      
>> Do you see some NOTIFY requests going around? they are used during
>> attended transfer to inform on the new call state.....
>>
>>
>>    
> Nope, no NOTIFY requests.
>
> Well wat is ment was making asterisk dumb, and just let if forward a
> complete call.. so instead of doing a dial to an opensips extention, just
> make a full transfer of the call to the opensips server, and then to the
> extention.
>
> I'm trying it the other way arround now, as you said earlier that the
> opensips recieves all the calls (so is directly connected to the sip trunk)
> but i have some strange issue's with that 2, i can't call outside and when i
> call inside, the phone rings (i just made a alias) and then i can't pick it
> up or anything, the phone doesn't respond!
>  
Hi Peter,

have you checked the SIP trace to see why the call is not established
when you pick up the ringing phone ?

Regards,
Bogdan

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