Voice problem with SIP trunk connection

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Apr 26th, 2010
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hI,


I am using a UC500 for one of our clients, we have 2 x BRI + 1 x SIP Trunk.

We use BRI tunks for incoming calls. --> No problem work fine.

We just configure a SIP Trunk for outgoing calls --> problem.


When we try to call an external number, connection is established correctly, we can hear correctly the conversation but the other party can't hear us.


You can find the config in attachement.


Do you have any idee ?


We are working with the same config on other site without any problem.


Also another problem that i can't understand is that output of sh sip regis statu :


Line                              peer        expires(sec)  registered
================================  ==========  ============  ==========
xxxxxxxxx                    20007       269           yes
xxxxxxxx                      20010       255           yes
201                               20006       256           yes
202                               20008       255           yes
203                               20009       213           yes
204                               20011       272           yes
205                               20012       261           yes
207                               20014       1             yes
208                               20015       262           yes
210                               20017       268           yes
211                               20018       85            yes
212                               20019       255           yes
213                               20020       92            yes
214                               20021       255           yes
400                               20026       251           yes
401                               20027       104           yes
402                               20028       262           yes
403                               20029       276           yes
404                               20030       278           yes
405                               20031       262           yes
406                               20032       275           yes
407                               20033       266           yes
408                               20034       277           yes
409                               20035       281           yes
410                               20036       217           yes
411                               20037       185           yes
412                               20038       285           yes


We must only see the SIP Trunk, why can i see the ip phone extension also ?


Regards,


Mohamed

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David Trad Mon, 04/26/2010 - 15:07
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Hi Mohamed,


A couple of things I would like for you to look at, in particular the following:


voice class codec 1
codec preference 1 g711ulaw

I noticed in your config the above was configured with only G711uLaw, i am just assuming here, but it looks as though you are wanting to force this particular Codec only right?


If not, then can you change it to the following please:

voice class codec 1

codec preference 1 g729r8
codec preference 2 g711ulaw
codec preference 3 g711alaw


The above however would need to be applied to your VoIP dial-peers which by looking at your config, the class codec already is.



Secondly I noticed you have no transcoding enabled, whilst in quite a few cases this is not needed, I have made it a habit myself to add it into the system, if you are using CCA to configure the system, which by the looks of it most of it was configured with CCA, or a CCA template, then use CCA to add transcoding in as you may find this might actually be needed with your SIP trunk depending on what interaction is happening between the system and your SIP trunk provider.


Manual transcoding can be applied by using the following code (NOTE: This code will not be within OOB guide lines, if you want it within OOB guidelines then please modify to operate within the guidelines specified in the OOB documenation).


sccp local (Interface or BVI)
sccp ccm 10.1.2.1 identifier 1 version 7.0
sccp
!
sccp ccm group 1
bind interface XXXXXXXXXXXXX
associate ccm 1 priority 1
associate profile 1 register transcode



dspfarm profile 1 transcode 
codec g711ulaw
codec g711alaw
codec g729ar8
codec g729abr8
codec g729r8
codec g729br8
maximum sessions 2   <---------------- Put there however many sessions you need for transcoding
associate application SCCP


***Telephony-service config****


sdspfarm units 2
sdspfarm transcode sessions 2
sdspfarm tag 2 transcode



In practice if you are going to continue using CCA to manage the system, then you should really use CCA to enable it, however granted that at times access to the system remotely using CCA is not always workable or can be done, so the above code is only if you are needing to do it manually.



As for the SIP registration status, this is quite normal if you have not done the following:

  1. You need to apply the following on all DN's "number 206no-reg primary" the no reg primary will prevent this from carrying out SIP registration, however do not apply this to your SIP trunk registration DN if you use one.
  2. Also make sure all your POTS dial-peers also have the "no sip-register" command, by the looks of it that is already applied, but it is good to give it a once over and make sure that is the case.




Right now this is all I can think of, I hope it helps you to resolve the problem one way or another :)



Cheers,



David.

mCiscotelecom Mon, 04/26/2010 - 23:44
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David,


Thanks you very much for all these remarques, i will implemente all these issues and coming back to you for news.


Thanks


Mohamed

Paul Cobley Tue, 04/27/2010 - 01:43
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Mohamed,


I had the same issue - resolved by binding the audio to the WAN interface instead of the loopback.


Best wishes,


Paul.

mCiscotelecom Tue, 04/27/2010 - 05:17
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Hello Paul,


And at which layer do you change it ?

Do you have the command line ?


Thanks in advance


Mohamed

Paul Cobley Tue, 04/27/2010 - 05:19
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voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
allow-connections sip to sip
  no supplementary-service sip moved-temporarily
no supplementary-service sip refer
sip
  bind control source-interface Loopback0
  bind media source-interface Loopback0 <= CHANGE THIS TO YOUR WAN INTERFACE

Maulik Shah Wed, 04/28/2010 - 11:01
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Mohamed

  In addition to the awesome tips posted already - can you let us know if there is any firewall in front of the UC500? We have seen this in the past with SIP trunk traffic not being allowed through a firewall.

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