HI Guys
I need help once again with a SIP setup.
Release L2.300.25 on OXE.
Client bought a third party call center solution call Real Connect. Currently 5 agents on PC's.
They asked me for a SIP trunk to the system. I set up the external gateway and SIP trunk with a network number 6005 (Also a DDI number in the system)
to route the calls to the Real Connect
- Firstly I set up the Trunkgroup as SIP ABCF. When dialling the the DDI number it routes from PRI via OXE to their server and IVR takes the call from there. Works 100%
But then we tested the call when dialling the main number and switchboard receives it and tries to transfer call to 6005. Call goes through and you start hearing the IVR and then call cuts. This happens from any internal extension as well. Call cuts few seconds after the transfer. Even when dialling from extension to extension and then transferring. (I thought it might be trunk to trunk problem but obviously its not.)
- Then I deleted the setup and created the SIP trunk as ISDN all Countries. Dial the DDI number from external or internal and it goes to IVR. Works 100%. When I dial another DDI number in the company and transfer the call to 6005 it goes through and works 100% without cutting the call when transferred. BUT...when trying to dial that network number 6005 from the switchboard (4059 IP), it gives me an error "INACCESSIBLE EXTENSION" Not 1 of the IP attendants can dial that network number. It works from IP phones.
I have attached my config of sip trunk and External sip gateway and a Motortrace3. Call cuts with external call from extension 6390 to 6005 (Network number) (PABX Room) when transferred.
thanx
Martin
Calls drop on transfer
Calls drop on transfer
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Re: Calls drop on transfer
hi MartinB
how solve this issue. i also have this issue. external call "T2- E1" incoming to IVR "sip device" . When IVR transfer call to PBX extn. the call is drop.
how solve this issue. i also have this issue. external call "T2- E1" incoming to IVR "sip device" . When IVR transfer call to PBX extn. the call is drop.
- cavagnaro
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Re: Calls drop on transfer
Please, post your logs and scenario description. The issue can be many things. Sip scenarios are more complex than tdm usually
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Re: Calls drop on transfer
hi Cavagnaro
thanks. And attached the log file.
when public call in then route to 43671. 43671 is the sip device to IVR. when IVR transfer to pbx 150. the call drop. At the trace show 503 Service Unavailable.
thanks. And attached the log file.
when public call in then route to 43671. 43671 is the sip device to IVR. when IVR transfer to pbx 150. the call drop. At the trace show 503 Service Unavailable.
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- cavagnaro
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Re: Calls drop on transfer
A 5XX error means there is bigger error, a system one, not just a bad configuration. Can be overload or something like that.
Better open a direct eSR for better support
Better open a direct eSR for better support
Ignorance is not the problem, the problem is the one who doesn't want to learn
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Re: Calls drop on transfer
Hi Guys
Sorry for only responding to this post now.
What I did was set up an ARS route and created Prefix (ARS prof trk group seizure with overlapping)
Created new Discriminator and pointed the ARS route to that discriminator and in the discriminator created the first digit to be used and send 4 digits.
So the Attendant seizes the trunk and then dials the Network number and call goes through fine.
It is only a workaround to the actual solution needed. Because the CLi of external call does not get passed through. So it always shows that the attendant is making a call to the Realconnect call center.
I
Sorry for only responding to this post now.
What I did was set up an ARS route and created Prefix (ARS prof trk group seizure with overlapping)
Created new Discriminator and pointed the ARS route to that discriminator and in the discriminator created the first digit to be used and send 4 digits.
So the Attendant seizes the trunk and then dials the Network number and call goes through fine.
It is only a workaround to the actual solution needed. Because the CLi of external call does not get passed through. So it always shows that the attendant is making a call to the Realconnect call center.
I