cannot find a suitable SIP URI to dial out | SIP Error

[RizwanDarvesh]

Hello

I am facing an issue with the call coming from a call center to Avaya IP Office (where my SIP trunk is connected). I have a shortcode 7N / Dial / N / LINE ID

My problem is that Avaya is literally forwarding the letter "N" to the SIP trunk.

If I add any literal number like 05XXXXXXXXXX, the exact 05XXXXXXXXXX is dialed, but in my case, since I will be dialing out to the PSTN, I can be the rest of the world, so I cannot add the exact numbers. But the shortcodes are also not working.

Here is the log, where you can see that the letter "N" is being transferred. If I remove any specific number, I also see a "cannot find a suitable SIP URI to dial out."

Please help me figure out how I can resolve this.



1228089mS Sip: SIP Line (18): Make Target voip, line group id is 17 and ip 192.168.42.100
1228089mS Sip: SIP Line (18) cannot find a suitable SIP URI to dial out
1228089mS Sip: SIP Line (17): Make Target voip, line group id is 17 and ip 192.168.3.1
1228090mS Sip: SIP Line (17): License, Valid 1, Available 10, Consumed 1
1228090mS Sip: SIP Line (17): sip_trunk_config_items 50004001, sip_trunk_config_items_2 00000000, voip.flags 00400949
1228090mS Sip: SIPDialog f1819300 created, dialogs 11 txn_keys 1 video 1 presentation 1 camera 1 unsupp audio 0
1228090mS Sip: 0000000000000000 0.1020.0 -1 SIPTrunk Endpoint(f1819300) SetUnIntTransactionCondition to UnInt_None
1228096mS Sip: c0a82a01000003fa 18.1018.1 5 SIPTrunk Endpoint(f4c2e620) received CMProceeding
1228096mS Sip: c0a82a01000003fa 18.1018.1 5 SIPTrunk Endpoint(f4c2c1bc) SIPEndPoint: Received an CMProceeding State Transition to SIPDialog::INVITE_RCVD(9)
1228099mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f181af50) received CMSetup
1228099mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) CMSetup received, owner f181cd20, dialog f1819300, dialling N
1228099mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) OnCmMessage Setup is_call_diverted 0 is_call_twinned 0 is_vmail_leave 0 out device 5 in device 5 full_name_supp 1
1228099mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) OnCmMessage Setup responding sets caller
1228100mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) OnCmMessage Setup user <> orig_user <> orig_hg <>
1228100mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) Init data on sip trunk calling_user <> orig_calling_obj <> caller <024082002@192.168.42.100> orig_caller <>
1228100mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) OnCmMessage Setup calling_party number <024082002@192.168.42.100> name <024082002>
1228100mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) FillInternalData user <> hg <> number <024082002@192.168.42.100> restricted 0
1228100mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) mNearName set to <024082002@192.168.42.100> mPrivate 0
1228100mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) mNearAlias set to <024082002>
1228100mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) FillInternalData user <> hg <> number <024082002@192.168.42.100> restricted 0
1228100mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) mNearContact set to <024082002@192.168.42.100>
1228101mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) mNearContactAlias set to <024082002>
1228101mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) SIPDialog::BuildSDPFromFastStart
1228102mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) SetLocalRTPAddress to 192.168.3.2:46750
1228103mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) SdpClone
1228103mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) UpdateSDPState SIPDialog::IDLE(0) -> SIPDialog::OFFER_SENT(1)
1228103mS Sip: c0a82a01000003fc 17.1020.0 5 SIPTrunk Endpoint(f1819300) INVITE SENT TO 192.168.3.1:5060 (reg required 0 registered 0)
1228104mS SIP Call Tx: 17
INVITE sip:N@192.168.3.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.2:5060;rport;branch=z9hG4bK4ea4a77d148d86e9f45ae54c19a541c1
From: "024082002" <sip:024082002@192.168.3.1>;tag=a325de676eb51287
To: <sip:N@192.168.3.1>
Call-ID: c93d802284455599323e4f6ec35e41f3
CSeq: 1755500807 INVITE
Contact: "024082002" <sip:024082002@192.168.3.2:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
P-Early-Media: supported
User-Agent: IP Office 11.1.0.0.0 build 237
Content-Type: application/sdp
Content-Length: 296

v=0
o=UserA 2504560353 166202799 IN IP4 192.168.3.2
s=Session SDP
c=IN IP4 192.168.3.2
t=0 0
m=audio 46750 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
1228105mS SIP Tx: UDP 192.168.3.2:5060 -> 192.168.3.1:5060
INVITE sip:N@192.168.3.1 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.2:5060;rport;branch=z9hG4bK4ea4a77d148d86e9f45ae54c19a541c1
From: "024082002" <sip:024082002@192.168.3.1>;tag=a325de676eb51287
To: <sip:N@192.168.3.1>
Call-ID: c93d802284455599323e4f6ec35e41f3
CSeq: 1755500807 INVITE
Contact: "024082002" <sip:024082002@192.168.3.2:5060;transport=udp>
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,INFO,REFER,NOTIFY,UPDATE
Supported: timer
P-Early-Media: supported
User-Agent: IP Office 11.1.0.0.0 build 237
Content-Type: application/sdp
Content-Length: 296

v=0
o=UserA 2504560353 166202799 IN IP4 192.168.3.2
s=Session SDP
c=IN IP4 192.168.3.2
t=0 0
m=audio 46750 RTP/AVP 8 0 18 4 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16

 

[MikeeOK]

Where is this short code? User, System or ARS?

If its user or system is it pointing to ARS? What's the shot codes on ARS? If its in ARS what's the sort code pointing to it?

What's the 7 for?

I assume this is an R11.1 system?

And this relates to your other post? Is this why it didn't work in your other post?