Hi Giorgio,<br><br>So it means that Asterisk has no native support for g729 ?<br><br>Thanks<br><br><div class="gmail_quote">On Wed, Mar 17, 2010 at 7:04 AM, Giorgio Incantalupo <span dir="ltr"><<a href="mailto:gincantalupo@fgasoftware.com">gincantalupo@fgasoftware.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Hi Bruno,<br>
<br>
I remember one of our customer had a similar problem with tellfree in<br>
Brazil. Their IT technician told me it was due to a g729 codec<br>
problem...they installed it and the problem disappeared. I never<br>
checked, I could only trust their man.<br>
Maybe it can help.<br>
<br>
Giorgio<br>
<br>
P.S.: let me know if it works, please!<br>
<div><div></div><div class="h5"><br>
Bruno Camargo wrote:<br>
> Hello Gentleman,<br>
><br>
> I'm new to asterisk, this is my first instalation, first post... so<br>
> I'd like to apologize if this question has already been made. I<br>
> googled but I couldn't find nothing similar.<br>
><br>
> Here's the thing.<br>
><br>
> I'm migrating from ATA to Asterisk one of my client's office and I<br>
> have a very simple setup.<br>
><br>
> A Linux PC running Debian Lenny + Asterisk 1.4.21. It's a totally<br>
> digital setup, it means I have no analogic cards connected.<br>
><br>
> I can make calls between my extension perfectly;<br>
> I can make outgoing calls without any problems;<br>
> Incoming calls are dropped after exatly 10 seconds; All incoming calls.<br>
><br>
> The asterisk box is hooked up to the LAN switch and it runs with a<br>
> private IP address. I have another Linux box performing<br>
> firewall/routing roles.<br>
><br>
> Outgoing and incoming calls working perfectly from the ATA (linksys<br>
> pap2t) but not from asterisk, because it hangs up after 10 seconds.<br>
><br>
> Some LOGS:<br>
><br>
> [Mar 16 15:11:12] DEBUG[13311] acl.c: ##### Testing 192.168.20.113<br>
> with 192.168.20.0<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 0: OPTIONS<br>
> sip:241@192.168.20.113:15956;rinstance=542e2865b2c6abe1 SIP/2.0 (71)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 1: Via: SIP/2.0/UDP<br>
> 192.168.20.249:5060;branch=z9hG4bK29f64fe1;rport (65)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 2: From: "asterisk"<br>
> <<a href="mailto:sip%3Aasterisk@192.168.20.249">sip:asterisk@192.168.20.249</a><br>
</div></div>> <mailto:<a href="mailto:sip%253Aasterisk@192.168.20.249">sip%3Aasterisk@192.168.20.249</a>>>;tag=as4bdc3589 (61)<br>
<div class="im">> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 3: To:<br>
> <sip:241@192.168.20.113:15956;rinstance=542e2865b2c6abe1> (61)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 4: Contact:<br>
</div>> <<a href="mailto:sip%3Aasterisk@192.168.20.249">sip:asterisk@192.168.20.249</a> <mailto:<a href="mailto:sip%253Aasterisk@192.168.20.249">sip%3Aasterisk@192.168.20.249</a>>> (38)<br>
<div class="im">> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 5: Call-ID:<br>
> <a href="mailto:7a4676c71af6501909db830431000932@192.168.20.249">7a4676c71af6501909db830431000932@192.168.20.249</a><br>
</div>> <mailto:<a href="mailto:7a4676c71af6501909db830431000932@192.168.20.249">7a4676c71af6501909db830431000932@192.168.20.249</a>> (56)<br>
<div class="im">> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 6: CSeq: 102 OPTIONS<br>
> (17)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 7: User-Agent:<br>
> Asterisk PBX (24)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 8: Max-Forwards: 70 (16)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 9: Date: Tue, 16 Mar<br>
> 2010 18:11:12 GMT (35)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 10: Allow: INVITE,<br>
> ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY (66)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 11: Supported:<br>
> replaces (19)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 12: Content-Length:<br>
> 0 (17)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: *** SIP TIMER: Initializing<br>
> retransmit timer on packet: Id #-1<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 0: SIP/2.0 200 OK (14)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 1: Via: SIP/2.0/UDP<br>
> 192.168.20.249:5060;branch=z9hG4bK29f64fe1;rport=5060 (70)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 2: Contact:<br>
</div>> <sip:<a href="http://192.168.20.113:15956" target="_blank">192.168.20.113:15956</a> <<a href="http://192.168.20.113:15956" target="_blank">http://192.168.20.113:15956</a>>> (35)<br>
<div class="im">> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 3: To:<br>
> <sip:241@192.168.20.113:15956;rinstance=542e2865b2c6abe1>;tag=67747e4a<br>
> (74)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 4: From:<br>
> "asterisk"<<a href="mailto:sip%3Aasterisk@192.168.20.249">sip:asterisk@192.168.20.249</a><br>
</div>> <mailto:<a href="mailto:sip%253Aasterisk@192.168.20.249">sip%3Aasterisk@192.168.20.249</a>>>;tag=as4bdc3589 (60)<br>
<div class="im">> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 5: Call-ID:<br>
> <a href="mailto:7a4676c71af6501909db830431000932@192.168.20.249">7a4676c71af6501909db830431000932@192.168.20.249</a><br>
</div>> <mailto:<a href="mailto:7a4676c71af6501909db830431000932@192.168.20.249">7a4676c71af6501909db830431000932@192.168.20.249</a>> (56)<br>
<div class="im">> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 6: CSeq: 102 OPTIONS<br>
> (17)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 7: Accept:<br>
> application/sdp (23)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 8: Accept-Language:<br>
> en (19)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 9: Allow: INVITE,<br>
> ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO (81)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 10: User-Agent:<br>
> X-Lite release 1104o stamp 56125 (44)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 11: Content-Length:<br>
> 0 (17)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Header 12: (0)<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: = Found Their Call ID:<br>
> <a href="mailto:7a4676c71af6501909db830431000932@192.168.20.249">7a4676c71af6501909db830431000932@192.168.20.249</a><br>
</div>> <mailto:<a href="mailto:7a4676c71af6501909db830431000932@192.168.20.249">7a4676c71af6501909db830431000932@192.168.20.249</a>> Their Tag<br>
<div class="im">> Our tag: as4bdc3589<br>
> [Mar 16 15:11:12] DEBUG[13311] chan_sip.c: ** SIP TIMER: Cancelling<br>
> retransmit of packet (reply received) Retransid #8282<br>
> *[Mar 16 15:11:12] DEBUG[13311] chan_sip.c: Stopping retransmission on<br>
> '<a href="mailto:7a4676c71af6501909db830431000932@192.168.20.249">7a4676c71af6501909db830431000932@192.168.20.249</a><br>
</div>> <mailto:<a href="mailto:7a4676c71af6501909db830431000932@192.168.20.249">7a4676c71af6501909db830431000932@192.168.20.249</a>>' of Request<br>
<div class="im">> 102: Match Found<br>
> [Mar 16 15:11:13] NOTICE[14413] rtp.c: Unknown RTP codec 126 received<br>
> from '192.168.20.113'<br>
> [Mar 16 15:11:13] WARNING[13311] chan_sip.c: Maximum retries exceeded<br>
> on transmission <a href="mailto:22021ea032130d5f3bd50ac67cf61e09@200.229.195.226">22021ea032130d5f3bd50ac67cf61e09@200.229.195.226</a><br>
</div>> <mailto:<a href="mailto:22021ea032130d5f3bd50ac67cf61e09@200.229.195.226">22021ea032130d5f3bd50ac67cf61e09@200.229.195.226</a>> for seqno<br>
<div class="im">> 102 (Critical Response)<br>
> [Mar 16 15:11:13] DEBUG[13311] chan_sip.c: Setting SIP_ALREADYGONE on<br>
> dialog <a href="mailto:22021ea032130d5f3bd50ac67cf61e09@200.229.195.226">22021ea032130d5f3bd50ac67cf61e09@200.229.195.226</a><br>
</div>> <mailto:<a href="mailto:22021ea032130d5f3bd50ac67cf61e09@200.229.195.226">22021ea032130d5f3bd50ac67cf61e09@200.229.195.226</a>><br>
<div class="im">> [Mar 16 15:11:13] WARNING[13311] chan_sip.c: Hanging up call<br>
> <a href="mailto:22021ea032130d5f3bd50ac67cf61e09@200.229.195.226">22021ea032130d5f3bd50ac67cf61e09@200.229.195.226</a><br>
</div>> <mailto:<a href="mailto:22021ea032130d5f3bd50ac67cf61e09@200.229.195.226">22021ea032130d5f3bd50ac67cf61e09@200.229.195.226</a>> - no reply<br>
<div class="im">> to our critical packet.<br>
> [Mar 16 15:11:13] DEBUG[14413] channel.c: Didn't get a frame from<br>
> channel: SIP/7977529-081d60d0<br>
> *[Mar 16 15:11:13] DEBUG[14413] channel.c: Bridge stops bridging<br>
> channels SIP/7977529-081d60d0 and SIP/241-081d7a50<br>
> [Mar 16 15:11:13] DEBUG[14413] channel.c: Hanging up channel<br>
> 'SIP/241-081d7a50'<br>
> [Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Hangup call<br>
> SIP/241-081d7a50, SIP callid<br>
> <a href="mailto:29d72fed0b17b16b76b12758136b3c25@192.168.20.249">29d72fed0b17b16b76b12758136b3c25@192.168.20.249</a><br>
</div>> <mailto:<a href="mailto:29d72fed0b17b16b76b12758136b3c25@192.168.20.249">29d72fed0b17b16b76b12758136b3c25@192.168.20.249</a>>)<br>
<div class="im">> [Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Strict routing enforced for<br>
> session <a href="mailto:29d72fed0b17b16b76b12758136b3c25@192.168.20.249">29d72fed0b17b16b76b12758136b3c25@192.168.20.249</a><br>
</div>> <mailto:<a href="mailto:29d72fed0b17b16b76b12758136b3c25@192.168.20.249">29d72fed0b17b16b76b12758136b3c25@192.168.20.249</a>><br>
<div class="im">> [Mar 16 15:11:13] DEBUG[14413] chan_sip.c: *** SIP TIMER: Initializing<br>
> retransmit timer on packet: Id #-1<br>
> [Mar 16 15:11:13] DEBUG[14413] devicestate.c: Notification of state<br>
> change to be queued on device/channel SIP/241-081d7a50<br>
> [Mar 16 15:11:13] DEBUG[14413] devicestate.c: Notification of state<br>
> change to be queued on device/channel SIP/241<br>
> [Mar 16 15:11:13] DEBUG[13304] devicestate.c: No provider found,<br>
> checking channel drivers for SIP - 241-081d7a50<br>
> [Mar 16 15:11:13] DEBUG[14413] rtp.c: Channel '<unspecified>' has no<br>
> RTP, not doing anything<br>
> [Mar 16 15:11:13] DEBUG[14413] app_dial.c: Exiting with DIALSTATUS=ANSWER.<br>
> [Mar 16 15:11:13] DEBUG[13304] chan_sip.c: Checking device state for<br>
> peer 241-081d7a50<br>
> [Mar 16 15:11:13] DEBUG[14413] pbx.c: Spawn extension<br>
> (incoming_calls,7977529,2) exited non-zero on 'SIP/7977529-081d60d0'<br>
> [Mar 16 15:11:13] DEBUG[14413] channel.c: Soft-Hanging up channel<br>
> 'SIP/7977529-081d60d0'<br>
> [Mar 16 15:11:13] DEBUG[14413] channel.c: Hanging up channel<br>
> 'SIP/7977529-081d60d0'<br>
> [Mar 16 15:11:13] DEBUG[14413] chan_sip.c: Hangup call<br>
> SIP/7977529-081d60d0, SIP callid<br>
> <a href="mailto:22021ea032130d5f3bd50ac67cf61e09@200.229.195.226">22021ea032130d5f3bd50ac67cf61e09@200.229.195.226</a><br>
</div>> <mailto:<a href="mailto:22021ea032130d5f3bd50ac67cf61e09@200.229.195.226">22021ea032130d5f3bd50ac67cf61e09@200.229.195.226</a>>)<br>
<div class="im">> [Mar 16 15:11:13] DEBUG[14413] devicestate.c: Notification of state<br>
> change to be queued on device/channel SIP/7977529-081d60d0<br>
> [Mar 16 15:11:13] DEBUG[14413] devicestate.c: Notification of state<br>
> change to be queued on device/channel SIP/7977529<br>
><br>
> #########################################<br>
><br>
> And now my extensions.conf and sip.conf<br>
><br>
> [general]<br>
> allowoverlap=no<br>
> allowguest=no<br>
> bindport=5060<br>
> bindaddr=0.0.0.0<br>
> externip=189.38.242.109<br>
</div>> localnet=<a href="http://192.168.20.0/255.255.255.0" target="_blank">192.168.20.0/255.255.255.0</a> <<a href="http://192.168.20.0/255.255.255.0" target="_blank">http://192.168.20.0/255.255.255.0</a>><br>
<div class="im">> srvlookup=yes<br>
> disallow=all<br>
> ;allow=g729<br>
> allow=ulaw<br>
> allow=alaw<br>
> tos_sip=cs3<br>
> tos_audio=ef<br>
> tos_video=af41<br>
> regcontext=incoming_calls<br>
> register=><br>
> 7977529@sip.tellfree.net:P<a href="http://ASSWD:7977529@sip.tellfree.net/7977529" target="_blank">ASSWD:7977529@sip.tellfree.net/7977529</a><br>
</div>> <<a href="http://ASSWD:7977529@sip.tellfree.net/7977529" target="_blank">http://ASSWD:7977529@sip.tellfree.net/7977529</a>><br>
<div class="im">><br>
> [tellfree]<br>
> type=friend<br>
> context=incoming_calls<br>
</div>> host=<a href="http://sip.tellfree.net" target="_blank">sip.tellfree.net</a> <<a href="http://sip.tellfree.net" target="_blank">http://sip.tellfree.net</a>><br>
<div class="im">> username=7977529<br>
> authuser=7977529<br>
> authname=7977529<br>
> secret=PASSWD<br>
</div>> Fromdomain=<a href="http://sip.tellfree.net" target="_blank">sip.tellfree.net</a> <<a href="http://sip.tellfree.net" target="_blank">http://sip.tellfree.net</a>><br>
<div><div></div><div class="h5">> fromuser=7977529<br>
> insecure=port,invite<br>
> qualify=yes<br>
> nat=yes<br>
> canreinvite=no<br>
><br>
> [xlite](!)<br>
> type=friend<br>
> host=dynamic<br>
> qualify=yes<br>
> context=phones<br>
> canreinvite=yes<br>
><br>
> [241](xlite)<br>
> username=241<br>
> callerid=241<br>
> secret=PASSWD_1<br>
><br>
> [242](xlite)<br>
> username=242<br>
> callerid=242<br>
> secret=PASSWD_2<br>
><br>
> [243](xlite)<br>
> username=243<br>
> callerid=243<br>
> secret=PASSWD_3<br>
><br>
> #############################################<br>
><br>
> [general]<br>
> autofallthrough=yes<br>
><br>
> [default]<br>
> exten => s,1,Verbose(1|Unrouted call handler)<br>
> exten => s,n,Answer()<br>
> exten => s,n,Wait(1)<br>
> exten => s,n,Playback(tt-weasels)<br>
> exten => s,n,Hangup()<br>
><br>
> [incoming_calls]<br>
> ;exten => 7977529,1,NoOp()<br>
> ;exten => 7977529,n,Dial(SIP/241|SIP/243,30,Tt)<br>
> exten => 7977529,1,Dial(SIP/241&SIP/243,30,Tt)<br>
> ;exten => 7977529,n,Dial(SIP/243,30,Tt)<br>
> exten => 7977529,n,Hangup()<br>
><br>
> [outgoing_calls]<br>
> exten => _0X.,1,NoOp()<br>
> exten => _0X.,n,Dial(Sip/${EXTEN}@tellfree,30,Tt)<br>
> exten => _0X.,n,Hangup<br>
> exten => _7X.,1,NoOp()<br>
> exten => _7X.,n,Dial(Sip/${EXTEN}@tellfree,30,Tt)<br>
> exten => _7X.,n,Hangup<br>
><br>
> [internal]<br>
> exten => _24[1-9],1,Verbose(1|Estension ${EXTEN})<br>
> exten => _24[1-9],n,SayDigits(${EXTEN})<br>
> exten => _24[1-9],n,Dial(SIP/${EXTEN},20,r)<br>
> exten => _24[1-9],n,Hangup<br>
><br>
> [phones]<br>
> include => internal<br>
> include => outgoing_calls<br>
<br>
<br>
</div></div><div><div></div><div class="h5">--<br>
_____________________________________________________________________<br>
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</div></div></blockquote></div><br><br clear="all"><br>-- <br>BrCaBadT<br>