Re: [Asterisk-Users] Codec matching weirdness
Hi! A better option and one Asterisk desperately needs is some kind of --lint option, Which would check the config for errors and useless misspelled options. smile I personal find one or more typos or misspelling a month, On my PBXs. Yes, indeed, same for me. My advice is to always do an extensions reload and immediately check the /var/log/asterisk/messages, but that'll still not catch everything. Just found this which explained strange errors I saw for two months: exten = 123,4,Playback(some-sound)) Haha - stupid, ain't it? Cheers, Philipp ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec matching weirdness
A better option and one Asterisk desperately needs is some kind of --lint option, Which would check the config for errors and useless misspelled options. smile I personal find one or more typos or misspelling a month, On my PBXs. Eric Wieling wrote: Maybe someone will write a patch to print an error to the console if reinvite= is found in the config file.? On Sat, 2004-01-17 at 15:44, Olle E. Johansson wrote: Dustin Goodwin wrote: I did find something interesting. If you set reinvite=yes then * can setup the RTP stream so that it avoids the media proxy in the * box completely. I haven't tested to see if it changes anything. Can we please kill reinvite - it does not exist in the SIP channel as an option for anything. Period. There is an option called canreinvite that you can set to yes or no. Setting reinvite to anything will not change anything at all. However, setting canreinvite to something will change ASterisk's behaviour during a SIP call. It may also break your conversation if your SIP device does not support the SIP re-invite mechanism. Please read: http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+canreinvite for more information. /Olle PS. I know that the reinvite option is mentioned in many archived e-mails, which does not help at all. Please do not add any more messages with this option, as it will only confuse users. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Codec matching weirdness
I am experiencing a problem that from list archive it appears others are running into. When I dial from Cisco 7960 via the * to Free World Dialup destinations that supports G.729 the call fails. The major error from the debug log is Jan 15 00:11:14 NOTICE[22545]: channel.c:1481 ast_set_read_format: Unable to find a path from G729A to ULAW Jan 15 00:11:14 NOTICE[22545]: channel.c:1451 ast_set_write_format: Unable to find a path from ULAW to G729A Me too? I've been wondering the same thing. I asked before and didn't really get anywhere either. Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec matching weirdness
I did find something interesting. If you set reinvite=yes then * can setup the RTP stream so that it avoids the media proxy in the * box completely. I haven't tested to see if it changes anything. - Dustin - [EMAIL PROTECTED] wrote: I am experiencing a problem that from list archive it appears others are running into. When I dial from Cisco 7960 via the * to Free World Dialup destinations that supports G.729 the call fails. The major error from the debug log is Jan 15 00:11:14 NOTICE[22545]: channel.c:1481 ast_set_read_format: Unable to find a path from G729A to ULAW Jan 15 00:11:14 NOTICE[22545]: channel.c:1451 ast_set_write_format: Unable to find a path from ULAW to G729A Me too? I've been wondering the same thing. I asked before and didn't really get anywhere either. Kevin ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec matching weirdness
Dustin Goodwin wrote: I did find something interesting. If you set reinvite=yes then * can setup the RTP stream so that it avoids the media proxy in the * box completely. I haven't tested to see if it changes anything. Can we please kill reinvite - it does not exist in the SIP channel as an option for anything. Period. There is an option called canreinvite that you can set to yes or no. Setting reinvite to anything will not change anything at all. However, setting canreinvite to something will change ASterisk's behaviour during a SIP call. It may also break your conversation if your SIP device does not support the SIP re-invite mechanism. Please read: http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+canreinvite for more information. /Olle PS. I know that the reinvite option is mentioned in many archived e-mails, which does not help at all. Please do not add any more messages with this option, as it will only confuse users. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec matching weirdness
Maybe someone will write a patch to print an error to the console if reinvite= is found in the config file.? On Sat, 2004-01-17 at 15:44, Olle E. Johansson wrote: Dustin Goodwin wrote: I did find something interesting. If you set reinvite=yes then * can setup the RTP stream so that it avoids the media proxy in the * box completely. I haven't tested to see if it changes anything. Can we please kill reinvite - it does not exist in the SIP channel as an option for anything. Period. There is an option called canreinvite that you can set to yes or no. Setting reinvite to anything will not change anything at all. However, setting canreinvite to something will change ASterisk's behaviour during a SIP call. It may also break your conversation if your SIP device does not support the SIP re-invite mechanism. Please read: http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+canreinvite for more information. /Olle PS. I know that the reinvite option is mentioned in many archived e-mails, which does not help at all. Please do not add any more messages with this option, as it will only confuse users. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Go to http://www.digium.com/index.php?menu=documentation and look at the Unofficial Links section. This section has links to a wide variety of 3rd party Asterisk related pages. My page is the Asterisk Resource Pages. BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec matching weirdness
Can we please kill reinvite - it does not exist in the SIP channel as an option for anything. Period. There is an option called canreinvite that you can set to yes or no. Setting reinvite to anything will not change anything at all. Olle, I thought I was the only that was loosing it with folks interchanging the two options. ;) Rich ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec matching weirdness
On Saturday 17 January 2004 15:44, Olle E. Johansson wrote: Dustin Goodwin wrote: I did find something interesting. If you set reinvite=yes then * can setup the RTP stream so that it avoids the media proxy in the * box completely. I haven't tested to see if it changes anything. Can we please kill reinvite - it does not exist in the SIP channel as an option for anything. Period. Bugnote 873 -Tilghman ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
