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06.17.2007 at 06:10PM PDT, ID: 22639652
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How do I eliminate these calls dropped by asterisk?

Asked by richardsimnett in IP Telephony, Voice Over IP

Tags: asterisk, path, codec

Hello,
can someone help me straighten out these asterisk errors I am having with my provider. I have all my sound files in GSM format, and when making calls I get these errors randomly. I'd like to eliminate this issue altogether.

 Oooh, we need to change our formats since our peer supports only 0x100 (g729) and not 0x4 (ulaw)
Jun 17 21:56:09 WARNING[10186] channel.c: Unable to find a codec translation path from g729 to slin
Jun 17 21:56:09 WARNING[10186] channel.c: Unable to find a codec translation path from g729 to slin
Jun 17 21:56:09 VERBOSE[11093] logger.c:     -- SIP/callcentric-0966ea10 is making progress passing it to Local/wtn@default-cf06,2
Jun 17 21:56:10 DEBUG[10186] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5d86e3fb5839d3764000650d746a018b@callcentric.com' Request 103: Found
Jun 17 21:56:10 VERBOSE[11093] logger.c:     -- SIP/callcentric-0966ea10 is making progress passing it to Local/wtn@default-cf06,2
Jun 17 21:56:11 DEBUG[10186] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5d86e3fb5839d3764000650d746a018b@callcentric.com' Request 103: Found
Jun 17 21:56:11 VERBOSE[11093] logger.c:     -- SIP/callcentric-0966ea10 is making progress passing it to Local/wtn@default-cf06,2
Jun 17 21:56:13 DEBUG[10186] chan_sip.c: (Provisional) Stopping retransmission (but retaining packet) on '5d86e3fb5839d3764000650d746a018b@callcentric.com' Request 103: Found
Jun 17 21:56:13 VERBOSE[11093] logger.c:     -- SIP/callcentric-0966ea10 is making progress passing it to Local/wtn@default-cf06,2
Jun 17 21:56:13 DEBUG[10186] chan_sip.c: Acked pending invite 103
Jun 17 21:56:13 DEBUG[10186] chan_sip.c: Stopping retransmission on '5d86e3fb5839d3764000650d746a018b@callcentric.com' of Request 103: Match Found
Jun 17 21:56:13 DEBUG[10186] chan_sip.c: build_route: Contact hop: <sip:4613bd2d282ca374526eb3788a9cb5b4@204.11.192.23:5060;transport=udp>
Jun 17 21:56:13 VERBOSE[11093] logger.c:     -- SIP/callcentric-0966ea10 answered Local/wtn@default-cf06,2
Jun 17 21:56:13 WARNING[11093] channel.c: No path to translate from Local/wtn@default-cf06,2(64) to SIP/callcentric-0966ea10(256)
Jun 17 21:56:13 WARNING[11093] app_dial.c: Had to drop call because I couldn't make Local/wtn@default-cf06,2 compatible with SIP/callcentric-0966ea10
Jun 17 21:56:13 DEBUG[11093] chan_sip.c: update_call_counter(15613067787) - decrement call limit counter
Jun 17 21:56:13 DEBUG[10175] channel.c: Avoiding initial deadlock for 'SIP/callcentric-0966ea10'
Jun 17 21:56:13 VERBOSE[11093] logger.c:   == Spawn extension (default, wtn, 1) exited non-zero on 'Local/wtn@default-cf06,2'

What do I need to do?

Worth 500 points.

Thanks,
RickStart Free Trial
[+][-]06.18.2007 at 01:02AM PDT, ID: 19305331

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About this solution

Zones: IP Telephony, Voice Over IP
Tags: asterisk, path, codec
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Solution Provided By: grblades
Participating Experts: 1
Solution Grade: A
 
 
 
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