10-03-2007 10:29 AM - edited 03-14-2019 11:53 PM
Hi,
When i try to connect to a PSTN Phone behind PBX which is connected to my VoIP Gateway, i don't get a ring back.
In the debug i see PI value 8 from the Router but i don't get SIP 180 Ringing. Instead i get SIP 183 SESSION PROGRESS.
How to get the 180 RINGING ??
Rgds
Sarva
10-09-2007 12:58 PM
Add this hidden command on the pots dial peer where the calls are going out of dial-peer:
progress_ind alert strip 8
01-18-2012 09:31 PM
check out this doc
01-18-2012 09:56 PM
Hi Mark
Excellent document ! Thanks for sharing.
Regards
Lavanya
03-08-2012 04:52 AM
Hi,
I have same problem.
GW receives Alerting from PSTN via ISDN PRI, but sends 183 Proceeding to CUCM 8.5 instead 180 Ringing. IP phone than display CallProcceding instead Alerting message.
Regards,
Jaroslav
03-08-2012 09:27 AM
03-08-2012 09:54 AM
Hi Mark,
I read document, but I need modify incoming 183 message, not outgoing. I look for solution, which is independent on CUCM version. I have same problem with one installation CUCM 7.1 and there aren’t these possibilities. So I need modify 183 on Cisco ISR. It's possible with sip-profiles?
03-08-2012 10:32 AM
Yes,
My apologies I think this is what your looking for
03-08-2012 10:52 AM
Hi,
I read this document, but it looks, that with sip-profile it’s possible modify only message content, but I need modify message code (message type).
On which depends 180 and 183 messages? When GW sends 180 and when 183 message? I don’t understand, why GW sends 183 Proceeding message, if I see in “isdn q931” debug, that GW received ALERTING via ISDN PRI.
03-08-2012 01:19 PM
Here is one method I just tested .
dial-peer voice 22 pots description Carrier Side application session destination-pattern 22.. progress_ind setup enable 3 progress_ind alert enable 8 progress_ind progress enable 8 direct-inward-dial port 1/0:15 forward-digits all dial-peer voice 34 voip description Inside destination-pattern 34.. progress_ind setup enable 3 progress_ind alert enable 8 progress_ind progress enable 8 session target ipv4:10.96.2.130 dtmf-relay h245-signal no call fallback
03-08-2012 01:37 PM
One other suggestion I read was to dissable media cut through which should send the 180.
Router(config-sip-ua)# disable-early-media 180
03-08-2012 10:53 PM
Hi Mark,
Your configuration is for H323. I tried this configuration with progress indicators for SIP (session protocol sipv2) and behavior is same – GW receives alerting from ISDN PRI and sends 183 Progress message instead 180 Ringing message
debug isdn q931
debug ccsip messages
Mar 9 06:41:26.659: ISDN Se0/0/0:15 Q931: TX -> SETUP pd = 8 callref = 0x07CA
Bearer Capability i = 0x8090A3
Standard = CCITT
Transfer Capability = Speech
Transfer Mode = Circuit
Transfer Rate = 64 kbit/s
Channel ID i = 0xA9839F
Exclusive, Channel 31
Progress Ind i = 0x8183 - Origination address is non-ISDN
Calling Party Number i = 0x0080, '553287'
Plan:Unknown, Type:Unknown
Called Party Number i = 0x80, '279538'
Plan:Unknown, Type:Unknown
Mar 9 06:41:26.659: //15369/B8D23DD08892/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.31.240.5:5060;branch=z9hG4bK421764
From: <553287>;tag=BEEBD8A0-C95553287>
To: <279538>279538>
Date: Fri, 09 Mar 2012 06:41:26 GMT
Call-ID: BAC003E2-68E911E1-889781D4-A11A5281@172.31.240.5
Timestamp: 1331275286
CSeq: 101 INVITE
Allow-Events: telephone-event
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
Mar 9 06:41:26.687: ISDN Se0/0/0:15 Q931: RX <- CALL_PROC pd = 8 callref = 0x8
7CA
Channel ID i = 0xA9839F
Exclusive, Channel 31
Mar 9 06:41:27.551: ISDN Se0/0/0:15 Q931: RX <- ALERTING pd = 8 callref = 0x87
CA
Progress Ind i = 0x8082 - Destination address is non-ISDN
Locking Shift to Codeset 5
Codeset 5 IE 0x32 i = 0x81
Mar 9 06:41:27.559: //15369/B8D23DD08892/SIP/Msg/ccsipDisplayMsg:
Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.31.240.5:5060;branch=z9hG4bK421764
From: <553287>;tag=BEEBD8A0-C95553287>
To: <279538>;tag=241DE260-1DC279538>
Date: Fri, 09 Mar 2012 06:41:26 GMT
Call-ID: BAC003E2-68E911E1-889781D4-A11A5281@172.31.240.5
Timestamp: 1331275286
CSeq: 101 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIF
Y, INFO, REGISTER
Allow-Events: telephone-event
Remote-Party-ID: <279538>;party=called279538>
;screen=no;privacy=off
Contact: <279538>279538>
Supported: sdp-anat
Server: Cisco-SIPGateway/IOS-12.x
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 194
v=0
o=CiscoSystemsSIP-GW-UserAgent 2312 2271 IN IP4 10.0.128.16
s=SIP Call
c=IN IP4 10.0.128.16
t=0 0
m=audio 17884 RTP/AVP 8
c=IN IP4 10.0.128.16
a=rtpmap:8 PCMA/8000
a=ptime:20
01-07-2013 05:00 AM
Hi Jaroslav,
I hit the same problem. Trace from my GW is exact same respect to yours.
I'd like to ask you if you finally came up with a solution, or not.
Thanks and best regards,
Andrea
01-31-2013 12:04 AM
Hi Andrea,
Unfortunately no. I used H323 instead SIP and here isn’t problem with ringing.
02-01-2013 12:56 AM
Hi Jaroslav,
we solved it at the end. Guys from the PBX managed to change configuration to send out to VGW a PI=8, and play the ringback accordingly.
Thanks
Andrea
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