Unfortunately not, I already tried different forms callerid(num). Always the same error.<br><br>I came across this entry in asterisk changelogs - maybe an update of asterisk will help.<br><br><pre>* Asterisk 1.4.36-rc1 Released.
2010-08-20 16:46 +0000 [r283048-283123] Richard Mudgett <<a href="mailto:rmudgett@digium.com">rmudgett@digium.com</a>>
        * channels/chan_dahdi.c: Merged revision 278274 from
         <a href="https://origsvn.digium.com/svn/asterisk/trunk">https://origsvn.digium.com/svn/asterisk/trunk</a> .......... r278274
         | rmudgett | 2010-07-20 17:38:13 -0500 (Tue, 20 Jul 2010) | 1
         line Reference correct struct member for unlikely event
         PRI_EVENT_CONFIG_ERR. ..........
        * channels/chan_dahdi.c: Q931 - Sending PROGRESS after sending
         ALERTING is a protocol error The PRI layer in chan_dadhi will
         check if a PROGRESS message has already been sent, and not allow
         sending another (although that is technically allowed by the Q931
         spec), however it does not protect against sending an ALERTING
         and then sending a PROGRESS message, which is a violation of the
         specification. Most switches don't seem to care too deeply about
         this, but some do, and will disconnect the call when receiving
         this invalid sequence. Protocol specification reference:
         T-REC-Q.931-199805-I page 223, "Figure A.5/Q.931 -- Overview
         protocol control (network side) point-point (sheet 3 of 8)"
         (closes issue #17874) Reported by: nic_bellamy Patches:
         asterisk-1.4-r282537_no-progress-after-alerting.patch uploaded by
         nic bellamy (license 299)
         asterisk-1.6.2-r282537_no-progress-after-alerting.patch uploaded
         by nic bellamy (license 299)
         asterisk-trunk-r282537_no-progress-after-alerting.patch uploaded
         by nic bellamy (license 299)</pre><br>thx<br>christian<br><br><div class="gmail_quote">On 18 July 2012 13:07, Mitul Limbani <span dir="ltr"><<a href="mailto:mitul@enterux.in" target="_blank">mitul@enterux.in</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Mebe your operator doesnt like the CallerID(num) set as NULL just remove the callerid(num) statement and let the standard callerId get set by network.<div>
<br clear="all">Regards,<br>Mitul Limbani,<br>Chief Architech & Founder,<br>
Enterux Solutions Pvt. Ltd.<br>110 Reena Complex, Opp. Nathani Steel, <br>Vidyavihar (W), Mumbai - 400 086. India<br><a href="http://www.enterux.com/" target="_blank">http://www.enterux.com/</a><br><a href="http://www.entvoice.com/" target="_blank">http://www.entvoice.com/</a><br>
email: <a href="mailto:mitul@enterux.in" target="_blank">mitul@enterux.in</a><br>DID: <a href="tel:%2B91-22-61447605" value="+912261447605" target="_blank">+91-22-61447605</a><br>Cell: <a href="tel:%2B91-9820332422" value="+919820332422" target="_blank">+91-9820332422</a><br>
<br><br>
<br><br><div class="gmail_quote"><div><div class="h5">On Wed, Jul 18, 2012 at 4:23 PM, Christian Gansberger <span dir="ltr"><<a href="mailto:christian.gansberger@accm.at" target="_blank">christian.gansberger@accm.at</a>></span> wrote:<br>
</div></div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div class="h5">Hi List!<br><br>I have a Problem with Telecom Hungary, if I set a callforwarding on the Snom, to an external number (mobile).<br>
Versions: Asterisk version 1.4.35, libpri 1.4.11.4, dahdi 2.6.0, snom-7.7.30<br><br>When I call the Snom (Extension 68), it responds with "302 moved temporarily", and Asterisk try to dial out over the LOCAL channel using DAHDI.<br>
I get a Congestion back from Telecom. Channel 0/2, span 1 got hangup request, cause 21<br><br><br>Here is cli output: <br><br> -- Accepting call from 'callerid' to '68' on channel 0/1, span 1<br> -- Executing [s@macro-station-fallback-Q-VM:5] Dial("DAHDI/1-1", "SIP/68|15|tTW") in new stack<br>
-- Called 68<br><br> -- Got SIP response 302 "Moved Temporarily" back from 10.70.x.xxx<br><br> -- Now forwarding DAHDI/1-1 to 'Local/*1mobilenr@snom68' (thanks to SIP/68-000076b8)<br> -- Executing [*1mobilenr@snom68:1] Macro("Local/*1mobilenr@snom68-2fe3,2", "dialout-dahdi-test|mobilenr|g1|") in new stack<br>
-- Executing [s@macro-dialout-dahdi-test:1] Set("Local/*1mobilenr@snom68-2fe3,2", "CALLERID(number)=") in new stack<br> -- Executing [s@macro-dialout-dahdi-test:2] Dial("Local/*1mobilenr@snom68-2fe3,2", "DAHDI/g1/mobilenr||") in new stack<br>
-- Requested transfer capability: 0x00 - SPEECH<br> -- Called g1/mobilnr<br> -- DAHDI/2-1 is proceeding passing it to Local/*1mobilenr@snom68-2fe3,2<br><br> -- Local/*1mobilenr@snom68-2fe3,1 is proceeding passing it to DAHDI/1-1<br>
<br> -- Channel 0/2, span 1 got hangup request, cause 21<br><br> -- DAHDI/2-1 is circuit-busy<br><br> -- Hungup 'DAHDI/2-1'<br><br> == Everyone is busy/congested at this time (1:0/1/0)<br><br><br>I have also an output from "pri intense debug" - But I think the Telecom is just not accepting the outgoing call.<br>
What do you think?<br><br><br>thanks<br>yours<span><font color="#888888"><br>christian<br><br><br><br><br><br><br><br><br><br><br>
</font></span><br></div></div><span class="HOEnZb"><font color="#888888">--<br>
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