04-26-2010 12:45 PM - edited 03-21-2019 02:29 AM
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
04-26-2010 03:07 PM
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 transcodedspfarm 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:
Right now this is all I can think of, I hope it helps you to resolve the problem one way or another :)
Cheers,
David.
04-26-2010 11:44 PM
David,
Thanks you very much for all these remarques, i will implemente all these issues and coming back to you for news.
Thanks
Mohamed
04-27-2010 01:43 AM
Mohamed,
I had the same issue - resolved by binding the audio to the WAN interface instead of the loopback.
Best wishes,
Paul.
04-27-2010 05:17 AM
Hello Paul,
And at which layer do you change it ?
Do you have the command line ?
Thanks in advance
Mohamed
04-27-2010 05:19 AM
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
04-28-2010 11:01 AM
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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