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New Member

SIP ip phone issues

Hi,

My company has just moved from ISDN to SIP. Everything is working fine just few things are giving issues.

1. All ip phones can dial out and recieve incoming calls. No problem, however when  I press REDIAL button on ip phone the call is not connecting. The number that is displayed when presseing the REDIAL button is 08 XXXXXXX how do I remove 08 from start so that it can dial the number? I am usign CUCM 8.6.

2. Some of the cisco 8945 ip phones can not make landline calls they can make all other outgoing calls. Any ideas?

3. There are sone 7911 phones that can not make outgoing calls while others can?

when we were running ISDN everything was working normal. Now with new Telco and SIP things are working and some of them are not. Any suggestions would be really helpful.

Thanks

Regards

Rohit.

7 REPLIES
Silver

SIP ip phone issues

If you use CUBE (probably you do) configure translation profiles on inbound and/or outbound dial-peers.

Or see if you can fiddle around on the CUCM to get the correct calling/called numbers.

SIP providers normally want to have a specific calling and called party number layout.

In general, do debugs on the CUBE (deb ccsip messages) and see what is happening.

You might have a codec issue (SIP media unsupported) for some phones or wrong number layout

JH

SIP ip phone issues

Hi Rohit.

For first question , verify on CUCM or on CUBE verify wher you are applying 08 before the number.

Anyway to remove it from called number you can configure a voice translation-rule than voice translation-profile on cube.

Eg.

voice translation-rule 10

rule 1 /^08/ //

voice translation-profile out-strip-08

translate called 10

and apply this profile to your outgoing dial-peer

For second and third question, verify CSS configuration on Phone's DN

HTH

Regards

Carlo

Please rate all helpful posts

"The more you help the more you learn"

Please rate all helpful posts "The more you help the more you learn"
New Member

SIP ip phone issues

Hi Guys,

Thanks for all your help. I managed to fix issue 3 by upgrading the firmware of 7911 phones. Now coming back to issue 1 and 2, all the translation is done in CUCM 8.6 rather then on CUBE. I am pasting some output for some information. let me know what do u think about it. Thanks once again guys you have been really helpful.

sh dial-peer voice summary

dial-peer hunt 0

             AD                                    PRE PASS                OUT

TAG    TYPE  MIN  OPER PREFIX    DEST-PATTERN      FER THRU SESS-TARGET    STAT PORT    KEEPALIVE

11     voip  up   up                                0  syst

21     voip  up   up             08610362..         0  syst ipv4:11.214.2.11              active

22     voip  up   up             08610362..         1  syst ipv4:11.214.2.11              active

23     voip  up   up             0893640...         1  syst ipv4:11.214.2.11              active

24     voip  up   up             0893640...         2  syst ipv4:11.214.2.11              active

31     voip  up   up             .T                 0  syst ipv4:123.102.30.131

Kind Regards

Rohit.

New Member

Re: SIP ip phone issues

Hi Guys,

Some of the output from the debugs that I want to share are as under :

244609: Nov 11 00:24:06.458: //419321/5B7819B284A1/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x2BC16E78
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 0893312789
Called Number            : 0893640159
Source IP Address (Sig  ): 172.16.1.5
Destn SIP Req Addr:Port  : 11.214.2.11:5060
Destn SIP Resp Addr:Port : 11.214.2.11:5060
Destination Name         : 11.214.2.11

244610: Nov 11 00:24:06.458: //419321/5B7819B284A1/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 172.16.1.5
Source IP Port    (Media): 17796
Destn  IP Address (Media): 11.214.2.11

Destn  IP Port    (Media): 27164
Orig Destn IP Address:Port (Media): [ - ]:0

244611: Nov 11 00:24:06.458: //419321/5B7819B284A1/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 16
Disconnect Cause (SIP)   : 200

244612: Nov 11 00:24:10.518: //419327/30F354000000/SIP/Call/sipSPICallInfo:
The Call Setup Information is:
Call Control Block (CCB) : 0x2BC11488
State of The Call        : STATE_DEAD
TCP Sockets Used         : NO
Calling Number           : 0893640354
Called Number            : 92168484
Source IP Address (Sig  ): 182.16.7.5


Destn SIP Req Addr:Port  : 11.214.2.11:5060
Destn SIP Resp Addr:Port : 11.214.2.11:5060
Destination Name         : 10.212.1.10

244613: Nov 11 00:24:10.518: //419327/30F354000000/SIP/Call/sipSPIMediaCallInfo:
Number of Media Streams: 1
Media Stream             : 1
Negotiated Codec         : g711ulaw
Negotiated Codec Bytes   : 160
Nego. Codec payload      : 0 (tx), 0 (rx)
Negotiated Dtmf-relay    : 6
Dtmf-relay Payload       : 101 (tx), 101 (rx)
Source IP Address (Media): 182.16.7.5
Source IP Port    (Media): 19474
Destn  IP Address (Media): 172.16.103.52
Destn  IP Port    (Media): 18644
Orig Destn IP Address:Port (Media): [ - ]:0

244614: Nov 11 00:24:10.518: //419327/30F354000000/SIP/Call/sipSPICallInfo:
Disconnect Cause (CC)    : 16
Disconnect Cause (SIP)   : 487

I am getting few calls like that. Rest of the calls are coming ok. Could you please suggest whats wrong.

Thanks

Kind Regards

Rohit.

New Member

Re: SIP ip phone issues

Hi Guys,

Sorry to bother again, but I am just trying to find out why this is going on. Some of the phone not able to call mobile phones but they are able to call all other numbers. When I did the call Analyze this is what I found.

it is only inviting from CUCM to SIP TRUNK. Any ideas for this ????

Thanks

Kind Regards

Rohit.

SIP ip phone issues

Hi Rohit.

In your previous post you asked how to remove 08 from the biginning of dialed number but on your trace I see that you are still calling 08XXXX.

Have you applied a translation profile to your outgoing DP 31?

Can you please share a full debug ccsip messages during an outgoing call?

Thanks

Regards

Carlo

Please rate all helpful posts

"The more you help the more you learn"

Please rate all helpful posts "The more you help the more you learn"

Re: SIP ip phone issues

Rohit,

Look out on your route patterns, coz ISDN and SIP provider patterns of accepting numbering is differ.

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