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SIP 180 RINGING Issue

blrnetwork
Level 1
Level 1

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

14 Replies 14

hadbou
Level 5
Level 5

Add this hidden command on the pots dial peer where the calls are going out of dial-peer:

progress_ind alert strip 8

Mark Pareja
Level 1
Level 1

Hi Mark

Excellent document ! Thanks for sharing.

Regards

Lavanya

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

Jaroslav,

     Please read this document .

https://supportforums.cisco.com/docs/DOC-21998

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?

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.

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

One other suggestion I read was to dissable media cut through which should send the 180.

Router(config-sip-ua)# disable-early-media 180

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-C95

To: <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-C95

To: <279538>;tag=241DE260-1DC

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=called

;screen=no;privacy=off

Contact: <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

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

Hi Andrea,

Unfortunately no.  I used H323 instead SIP and here isn’t problem with ringing.

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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