: Tuesday, November 09, 2010 3:41 PM
To: asterisk-ss7@lists.digium.com
Subject: Re: [asterisk-ss7] No Audio on SS7 calls to Remote PRIs
Please past your chan_dahdi.conf and system.conf.
Also check if you have ulaw selected in your sip phone.
On Wed, Nov 10, 2010 at 12:35 AM, dave
;
>
> Thanks,
>
> Dave
>
>
>
> *From:* asterisk-ss7-boun...@lists.digium.com [mailto:
> asterisk-ss7-boun...@lists.digium.com] *On Behalf Of *dave george
> *Sent:* Tuesday, November 09, 2010 9:10 AM
> *To:* asterisk-ss7@lists.digium.com
> *Subject:* Re: [as
: asterisk-ss7@lists.digium.com
Subject: Re: [asterisk-ss7] No Audio on SS7 calls to Remote PRIs
Hi Guys,
I am having a similar issue with no audio. Other end is an ericsson switch.
See the logs below. I can make and receive calls fine but no audio. I
checked my CIC and they are lined up well
Hi Guys,
I am having a similar issue with no audio. Other end is an ericsson switch.
See the logs below. I can make and receive calls fine but no audio. I
checked my CIC and they are lined up well. I run dahdi_monitor but I only
see activity from my end (softphone registered on asterisk).
i m using asterisk 1.6.2.13 and working fine for me with dahdi-2.3.
On Thu, Oct 7, 2010 at 8:08 PM, Stephan Ellis wrote:
> It's working like a champ for me!
>
>
> On Wed, Oct 6, 2010 at 11:36 PM, bipin singh
> wrote:
>
>> Hi
>> use only asterisk-1.6.0 because it is working in my case
It's working like a champ for me!
On Wed, Oct 6, 2010 at 11:36 PM, bipin singh wrote:
> Hi
> use only asterisk-1.6.0 because it is working in my case .
>
>
> On Tue, Oct 5, 2010 at 7:32 PM, Stephan Ellis wrote:
>
>> Any specific point version of 1.6.0 i should use? Or just 1.6.0?
>>
>> -
Hi
use only asterisk-1.6.0 because it is working in my case .
On Tue, Oct 5, 2010 at 7:32 PM, Stephan Ellis wrote:
> Any specific point version of 1.6.0 i should use? Or just 1.6.0?
>
> -stephan
>
>
> On Tue, Oct 5, 2010 at 1:11 AM, bipin singh wrote:
>
>> Hi
>>Use asterisk-1.6.0 ver
On 10/05/2010 10:16 AM, Stephan Ellis wrote:
> Sorry, the link to Asterisk 1.6.0 is:
>
> http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.0.tar.gz
Yeah, I forgot that when we released that we hadn't yet switched to the
four-component version numbers. 1.6.0 is really 1.6.
Sorry, the link to Asterisk 1.6.0 is:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-1.6.0.tar.gz
-stephan
On Tue, Oct 5, 2010 at 9:46 AM, Stephan Ellis wrote:
> Switching to 1.6.0 did the trick. I tried to run 1.6.0.28 but I had the no
> audio issue. I'm not sure what
Switching to 1.6.0 did the trick. I tried to run 1.6.0.28 but I had the no
audio issue. I'm not sure what you mean by there is no 1.6.0. I found it
here:
http://downloads.asterisk.org/pub/telephony/asterisk/releases/asterisk-addons-1.6.0.tar.gz
You're welcome to get with me out of band to see m
On 10/05/2010 09:02 AM, Stephan Ellis wrote:
> Any specific point version of 1.6.0 i should use? Or just 1.6.0?
If the underlying problem is, as your switch technician suggests, lack
of response to a particular SS7 message, then going back to an older
version of Asterisk is not going to help. I kn
Any specific point version of 1.6.0 i should use? Or just 1.6.0?
-stephan
On Tue, Oct 5, 2010 at 1:11 AM, bipin singh wrote:
> Hi
>Use asterisk-1.6.0 version its work . Your configuration is ok .
>
> On Thu, Sep 30, 2010 at 7:45 PM, Stephan Ellis wrote:
>
>> All,
>>
>> I've got a problem o
Hi
Use asterisk-1.6.0 version its work . Your configuration is ok .
On Thu, Sep 30, 2010 at 7:45 PM, Stephan Ellis wrote:
> All,
>
> I've got a problem on my SS7 implementation. When I originate calls
> across my SS7 link and the call lands on a PRI, I get no audio in either
> direction. T
Actually CICs Start on channel 2, signalling is on channel 1. Any other
combination of cicstartswith results on no audio regardless of destination.
Remeber, it's only remote PRIs that I am having trouble with.
Also, if I set cicstartswith=2, when I issue "ss7 block linkset 1", the
switch only ack
are you sure that CIC start at channel 1?
have you tried changing values like 2 or 3?
Please past any GRS/GRA messages from asterisk cli.
On Mon, Oct 4, 2010 at 11:43 PM, Stephan Ellis wrote:
> I am definitely sure. Also, when starting asterisk on this box, it says:
>
> MTP2 link up (SLC 0)
>
I am definitely sure. Also, when starting asterisk on this box, it says:
MTP2 link up (SLC 0)
--- SS7 Up ---
Resetting CICs 1 to 23
Got reset acknowledgement from CIC 1 to 23.
So it looks like to two ends agree on the CIC mappings. It's weird because
it seems to only do this when calling remote
2010/10/4 Stephan Ellis :
> Anyone have any ideas on this?
>
> -stephan
>
As with most cases of no-audio in ss7:
cicbeginswith=1
channel=2-24
sigchan=1
you are 100% sure that you start numbering CICs with 1, and on the 2nd
one you put first audio channel?
please use debug on ss7, and restart yo
Anyone have any ideas on this?
-stephan
On Thu, Sep 30, 2010 at 10:30 AM, Stephan Ellis wrote:
> Yes that's correct. Sorry, I should be more clear about my setup. I work
> for a rural telephone company. We have our asterisk box connected to a
> Siemens EWSD. I have my softphone connected dir
Yes that's correct. Sorry, I should be more clear about my setup. I work
for a rural telephone company. We have our asterisk box connected to a
Siemens EWSD. I have my softphone connected directly to the asterisk box.
The box I am calling is an asterisk box connected to a PRI from bell. I get
just to clarify... you have the following setup: ss7 -> asterisk -> sip ->
softphone
where is the PRI ?
On Thu, Sep 30, 2010 at 11:04 AM, Stephan Ellis wrote:
> I do see audio being received, but I don't hear it on my softphone. I see
> no TX at all. Interestingly, the guy on the pri I was c
I do see audio being received, but I don't hear it on my softphone. I see
no TX at all. Interestingly, the guy on the pri I was calling said he could
hear me. The remote pri is an asterisk box, so i set a DID on it to go
straight to the echo test. While that system is playing demo-echo I see RX
Hi
Have you tried using dahdi_monitor to see if any sound is received ?
Rgds,
J.
On Thu, Sep 30, 2010 at 10:15 AM, Stephan Ellis wrote:
> All,
>
> I've got a problem on my SS7 implementation. When I originate calls
> across my SS7 link and the call lands on a PRI, I get no audio in either
>
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