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?

 

[RizwanDarvesh]

Mikee

It is a system shortcode, and it is pointing to an ARS. I have an SIP trunk connected to IP 500, which is being used by both the Avaya Extensions and call center agents.

For my Avaya extensions, the users press 9 to seize the trunk and then dial any number using the SIP trunk. The system shortcode for all of the users is "9N/N/Dial/line 17
For my call center agents, again, I have a different ARS with a shortcode "7N/N/Dial/Line 17." I am simply pointing the incoming calls from the call center to a shortcode "7N".

I am new to IPO 500. I am sorry if I have done something totally wrong, but I am using a shortcode of "9N" for the Avaya users and "7N" for call center agents. I have an incoming call route from the call center SIP trunk (line 18), and the destination is the shortcode "7N." I would like to mention that "9N" and "7N" use the same line 17 for outbound but in different ARS.

Yes, this is an R11.1 system. If I use "N" as the telephone number, I see the literal "N" or "X" in the invite being sent to the SIP trunk. If I replace "N" with an actual phone number like "7N/+123456789/Dial/Line 17," then the number "+123456789" is dialled successfully, and the end user will receive the call. However, since this is a call center, I cannot add all the numbers worldwide.

I need some logic regarding where Avaya will add the SIP URI of the person being called to the invite being sent to the SIP trunk.
I want a call from "Call Center > Avaya IPO > to go straight to SIP TRUNK."
Yes, I have the other post regarding the same issue. Calls from the SIP trunk to the call center are working fine, but calls from the call center to the SIP trunk using Avaya are not working.

 

[mhoxie]

You're need to use an ARS table instead of putting the call directly to the SIP URI line number.

 

[derfloh]

Even if I prefer ARS it should work without ARS by directly using the SIP trunk line group ID within short codes.

As long as you don’t slow your short codes, we will not find the reason.