Sip truniking and AA

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haroun
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Sip truniking and AA

Post by haroun »

Hi guys
getting some troubles making the Incoming call to AA through sip, when the AA acces number is called the attendent set is rong and the calling party is hearing the operator waiting guide , if routed to an extension calls worj fine.

VOIPProvider-----4g/lte-----Cisco-isr--------OXE--AA
in the sip trace we can clearly see that AA acces number is called .
any hint ?
thanx in advance
btw : the AA is working fine under PCMr2
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frank
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Re: Sip truniking and AA

Post by frank »

When you say it works when you route to an extension, are those SIP extensions also ? or Digital or IP ?
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haroun
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Re: Sip truniking and AA

Post by haroun »

hi frank
when i say it works is under pcmr2 trkgp the iaa is ansawring properly , the caller is able to hear the iaa voice guide prompt : welcome bla bala ....., and can dial digits .
for sip trunking when i route the incoming call :
1-to an extension via sda or num-exp every thing is ok
2-when the SDA or num-exp traget is the IAA the caller never get the IAA VOICE PROMPT instead he will hear the attandant's hold on voice guide(please hold an antandant will answer) and at sametime the call is ringing on the attandant console
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frank
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Re: Sip truniking and AA

Post by frank »

And the AA is what? 4645 or VAA ?
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haroun
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Re: Sip truniking and AA

Post by haroun »

the AA is the regular IAA Application
sadim
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Re: Sip truniking and AA

Post by sadim »

Hi,
Can you also attach the pcmr2 trunk grp configuration.
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haroun
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Re: Sip truniking and AA

Post by haroun »

hi @sadim here's bellow
btw a sip device connected to cisco (AS SERVER) get the same issue when call IAA
Node Number (reserved) : 1 x
x Trunk Group Id : 1800 x
x Instance (reserved) : 1 x
x x
x Trunk Group Type + MIXED x
x T2 Specificity + None x
x Public Network Ref. : 1 x
x VG for non-existent No. + YES x
x Entity Number : 0 x
x Supervised by Routing + NO x
x VPN Cost Limit for Incom.Calls : 0 x
x Immediat Trk Listening For VPN Cal + YES x
x VPN TS % : 50 x
x Csta Monitored + NO x
x Max.% of trunks : 0 x
x Ratio analog.to ISDN tax : ------ x
x TS Distribution on Accesses + YES x
x Quality Profile for Voice on IP + Profile #1 x
x IP compression type + Default x
x Use of volume in system + YES x
x Use of volume in system + YES x
x Announcement for Dialtone + NO x
x Announcement for Ringtone + NO x
x Private to Public Overflow + YES x
x Dialling end to end + NO x
x DTMF end to end signal. + NO x
x TS Permanently assigned + YES x
x Min. Nb.of digits on seize : 0 x
x Trunk group used in DISA + YES x
x DISA Secret Code : ---- x
x Routing To Executive + YES x
x Dissuasion For ACD + NO x
x Automated Attendant + YES x
x Calling party Rights category : 0 x
x Number To Be Added : -------- x
x Access Cluster Id : -1 x
x Collect Calls Allowed + YES x
x PCM Network Mode + NO x
x Barring mode + Not barred x
mqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqqq
sadim
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Re: Sip truniking and AA

Post by sadim »

Hi,
Seems to be ok.
Regarding the SIP, is it a SIP_ISDN or a SIP-ABC trunk (check trk parameter "Q931 signal variant")
haroun
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Re: Sip truniking and AA

Post by haroun »

yep it's SIP_ISDN, out /incomnig calls are working perfectly except when we use the IAA
sadim
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Re: Sip truniking and AA

Post by sadim »

For SIP-ISDN should work.
Perhaps a bug? R9.1 i1.605.41 is very old
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