11-09-2013 09:24 AM - edited 03-16-2019 08:19 PM
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.
11-10-2013 01:45 AM
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
11-10-2013 02:17 AM
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
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"The more you help the more you learn"
11-10-2013 07:26 PM
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.
11-10-2013 08:18 PM
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.
11-10-2013 08:58 PM
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.
11-11-2013 03:48 AM
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"
11-10-2013 10:58 PM
Rohit,
Look out on your route patterns, coz ISDN and SIP provider patterns of accepting numbering is differ.
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