[Asterisk-Users] chan_agent.c fails to compile
Using cvs head downloaded as of just a few minutes ago.. chan_agent.c: In function `action_agents': chan_agent.c:1446: warning: long int format, time_t arg (arg 7) chan_agent.c: In function `__login_exec': chan_agent.c:1684: syntax error before `char' chan_agent.c:1701: `agent_goodbye' undeclared (first use in this function) chan_agent.c:1701: (Each undeclared identifier is reported only once chan_agent.c:1701: for each function it appears in.) chan_agent.c:1708: `tmpoptions' undeclared (first use in this function) chan_agent.c:1714: `update_cdr' undeclared (first use in this function) chan_agent.c:1732: `context' undeclared (first use in this function) chan_agent.c:1737: `play_announcement' undeclared (first use in this function) chan_agent.c:1864: `filename' undeclared (first use in this function) gmake[1]: *** [chan_agent.o] Error 1 gmake[1]: Leaving directory `/usr/local/src/asterisk/channels' gmake: *** [subdirs] Error 1 I am feeling rather blind as I cannot see the issue.. can some kind person take a look? begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] chan_agent.c fails to compile
gcc version 2.95.3 20010125 (prerelease, propolice) on OpenBSD 3.6. BJ Weschke wrote: Compiled fine here. What version of GCC are you using? On 11/3/05, Matt Hess [EMAIL PROTECTED] wrote: Using cvs head downloaded as of just a few minutes ago.. chan_agent.c: In function `action_agents': chan_agent.c:1446: warning: long int format, time_t arg (arg 7) chan_agent.c: In function `__login_exec': chan_agent.c:1684: syntax error before `char' chan_agent.c:1701: `agent_goodbye' undeclared (first use in this function) chan_agent.c:1701: (Each undeclared identifier is reported only once chan_agent.c:1701: for each function it appears in.) chan_agent.c:1708: `tmpoptions' undeclared (first use in this function) chan_agent.c:1714: `update_cdr' undeclared (first use in this function) chan_agent.c:1732: `context' undeclared (first use in this function) chan_agent.c:1737: `play_announcement' undeclared (first use in this function) chan_agent.c:1864: `filename' undeclared (first use in this function) gmake[1]: *** [chan_agent.o] Error 1 gmake[1]: Leaving directory `/usr/local/src/asterisk/channels' gmake: *** [subdirs] Error 1 I am feeling rather blind as I cannot see the issue.. can some kind person take a look? ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] AudioCodes - TP260
We use tp-260 boards for ss7/sigtran.. they seem to behave similarly to mp2000 or tp1610 series boards which we have used with both mgcp and sip protocols.. their stuff seems to work rather well .. at least for us but YMMV. Chard Johnston wrote: Hi All, Does anyone have any experience with using Asterisk with AudioCodes TP260 SIP board? If yes, please let me know if you have had any problems. Regards, Chard Johnston ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] zaptel.conf config for CAS signalling
Well, I looked at the source and discovered I had to use a cas=1-24:1101 line to get the ztcfg to be happy.. now I'm moving on and wondering how do I specify cas signaling in the zapata.conf? Is that with the r2 option or something else? Humberto Aicardi wrote: Even better, share the whole zaptel.conf Humberto would you please share line 213 with us? On 10/18/05, Matt Hess [EMAIL PROTECTED] wrote: I have a customer that needs to do cas signaling across a t1,esf span.. it looks like this can be done but I'm not sure how as the documentation is very light in regards to cas.. it would appear that I need to use sf signaling but I get an error saying: $ ztcfg -vv Notice: Configuration file is /etc/zaptel.conf line 213: Unknown keyword 'sf' I've also tried the format suggested in zaptel.conf channel# = (etc.) but I continue to fail.. I'd love a few pointers here.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] setting a dialplan on a GXP-2000 Grandstream
Not that I've seen.. about all you can do is adjust the inter digit timeout.. Louis-David Mitterrand wrote: Hi, I looked at the docs and probably missed it: is there a way to set a dialplan on the GXP-2000? (to avoid having to press Send) Thanks, begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip rfc bye violated?
I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRelINVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx SIP/2.0 / 102 INVITE 11. CancelDestroy 12. Unhold SIP/2.0 13. TxReq ACK / 102 ACK 14. TxReqRelINVITE / 103 INVITE 15. Rx SIP/2.0 / 103 INVITE 16. CancelDestroy 17. Rx SIP/2.0 / 103 INVITE 18. CancelDestroy 19. Unhold SIP/2.0 20. TxReq ACK / 103 ACK 21. TxReqRelINVITE / 104 INVITE 22. Rx BYE / 302 BYE 23. TxResp SIP/2.0 / 302 BYE 24. Rx SIP/2.0 / 104 INVITE 25. CancelDestroy Why is asterisk allowing an invite after receiving a bye on a particular session/channel? From what I've read.. a bye should be the termination of the session/channel and therefore it should be hungup and removed.. yet it is not. I am using cvs head from 2005-10-08 00:00 .. I can't use the latest cvs head as it's rather ugly with sip right now.. especially on refer/redirect/reinvites.. but that will be left for a different topic. I believe from looking at things that the sip gateway involved with the sip session is re-using a particular call identifier immediately after it believes that call from before is gone.. (possibly a bug on the vendor side as far as that goes) but regardless of whether the vendor is immediately re-using a session id or not should not matter as the fact seems to be that asterisk allows this situation to happen when (from what I've been reading) it should not. Does anyone have any comments or thoughts on this? begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] sip rfc bye violated?
Attached is a pcap of sip packets that pertain to another call similar to the history shown.. it's hard to nail these down as it takes a lot of time, patience and sifting through dumps. Olle E. Johansson wrote: Matt Hess wrote: I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRelINVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx SIP/2.0 / 102 INVITE 11. CancelDestroy 12. Unhold SIP/2.0 13. TxReq ACK / 102 ACK 14. TxReqRelINVITE / 103 INVITE 15. Rx SIP/2.0 / 103 INVITE 16. CancelDestroy 17. Rx SIP/2.0 / 103 INVITE 18. CancelDestroy 19. Unhold SIP/2.0 20. TxReq ACK / 103 ACK 21. TxReqRelINVITE / 104 INVITE 22. Rx BYE / 302 BYE 23. TxResp SIP/2.0 / 302 BYE 24. Rx SIP/2.0 / 104 INVITE 25. CancelDestroy Why is asterisk allowing an invite after receiving a bye on a particular session/channel? From what I've read.. a bye should be the termination of the session/channel and therefore it should be hungup and removed.. yet it is not. I am using cvs head from 2005-10-08 00:00 .. I can't use the latest cvs head as it's rather ugly with sip right now.. especially on refer/redirect/reinvites.. but that will be left for a different topic. I believe from looking at things that the sip gateway involved with the sip session is re-using a particular call identifier immediately after it believes that call from before is gone.. (possibly a bug on the vendor side as far as that goes) but regardless of whether the vendor is immediately re-using a session id or not should not matter as the fact seems to be that asterisk allows this situation to happen when (from what I've been reading) it should not. Does anyone have any comments or thoughts on this? This history does not show the details on what Asterisk does. It seems like Asterisk transmits an INVITE, then gets a BYE and after the BYE get a response to the INVITE... Please provide a full SIP log so I see how we react to the response of the INVITE... /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users svi error.pcap Description: Binary data begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] sip rfc bye violated?
I should have mentioned that I can't do a full sip log.. with several calls a second whipping through this system it's almost impossible to weed out the info for the proper call.. and usually I don't see the dead channel until well after the fact. I did grab a fresh history and packet capture of a new(?) dead call. sip show history [EMAIL PROTECTED] tranquility*CLI * SIP Call 1. Rx INVITE / 1 INVITE 2. CancelDestroy 3. TxResp SIP/2.0 / 1 INVITE 4. TxResp SIP/2.0 / 1 INVITE 5. TxResp SIP/2.0 / 1 INVITE 6. TxRespRel SIP/2.0 / 1 INVITE 7. Rx ACK / 1 ACK 8. TxReqRelINVITE / 102 INVITE 9. Rx SIP/2.0 / 102 INVITE 10. CancelDestroy 11. Rx SIP/2.0 / 102 INVITE 12. CancelDestroy 13. Unhold SIP/2.0 14. TxReq ACK / 102 ACK 15. TxReqRelINVITE / 103 INVITE 16. Rx SIP/2.0 / 103 INVITE 17. CancelDestroy 18. Rx BYE / 201 BYE 19. TxResp SIP/2.0 / 201 BYE And the packet capture is attached again.. Matt Hess wrote: Attached is a pcap of sip packets that pertain to another call similar to the history shown.. it's hard to nail these down as it takes a lot of time, patience and sifting through dumps. Olle E. Johansson wrote: Matt Hess wrote: I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRelINVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx SIP/2.0 / 102 INVITE 11. CancelDestroy 12. Unhold SIP/2.0 13. TxReq ACK / 102 ACK 14. TxReqRelINVITE / 103 INVITE 15. Rx SIP/2.0 / 103 INVITE 16. CancelDestroy 17. Rx SIP/2.0 / 103 INVITE 18. CancelDestroy 19. Unhold SIP/2.0 20. TxReq ACK / 103 ACK 21. TxReqRelINVITE / 104 INVITE 22. Rx BYE / 302 BYE 23. TxResp SIP/2.0 / 302 BYE 24. Rx SIP/2.0 / 104 INVITE 25. CancelDestroy Why is asterisk allowing an invite after receiving a bye on a particular session/channel? From what I've read.. a bye should be the termination of the session/channel and therefore it should be hungup and removed.. yet it is not. I am using cvs head from 2005-10-08 00:00 .. I can't use the latest cvs head as it's rather ugly with sip right now.. especially on refer/redirect/reinvites.. but that will be left for a different topic. I believe from looking at things that the sip gateway involved with the sip session is re-using a particular call identifier immediately after it believes that call from before is gone.. (possibly a bug on the vendor side as far as that goes) but regardless of whether the vendor is immediately re-using a session id or not should not matter as the fact seems to be that asterisk allows this situation to happen when (from what I've been reading) it should not. Does anyone have any comments or thoughts on this? This history does not show the details on what Asterisk does. It seems like Asterisk transmits an INVITE, then gets a BYE and after the BYE get a response to the INVITE... Please provide a full SIP log so I see how we react to the response of the INVITE... /O ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users odd call.pcap Description: Binary data begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel.conf config for CAS signalling
I have a customer that needs to do cas signaling across a t1,esf span.. it looks like this can be done but I'm not sure how as the documentation is very light in regards to cas.. it would appear that I need to use sf signaling but I get an error saying: $ ztcfg -vv Notice: Configuration file is /etc/zaptel.conf line 213: Unknown keyword 'sf' I've also tried the format suggested in zaptel.conf channel# = (etc.) but I continue to fail.. I'd love a few pointers here.. begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] sip channels marked with SIP_NEEDDESTROY but not being removed
I have been seeing the subject behavior on head for a few days now.. (been trying nightly builds to see if a bug causing this has been fixed) on a sip show channels I get a little of active channels that I can correlate calls to.. but I also have some dead channels listed that should no longer be there but still are anyway.. in the sip show channels list these channels are marked with a (d).. in looking at chan_sip.c the channel should be marked as SIP_NEEDDESTROY and should be removed in looking at the source.. The history for such a channel looks like: tranquility*CLI sip show history -363845771@ tranquility*CLI * SIP Call 1. Rx INVITE / 1 INVITE 2. CancelDestroy 3. TxResp SIP/2.0 / 1 INVITE 4. TxResp SIP/2.0 / 1 INVITE 5. TxRespRel SIP/2.0 / 1 INVITE 6. Rx ACK / 1 ACK 7. TxReqRelINVITE / 102 INVITE 8. Rx SIP/2.0 / 102 INVITE 9. CancelDestroy 10. Rx SIP/2.0 / 102 INVITE 11. CancelDestroy 12. Unhold SIP/2.0 13. TxReq ACK / 102 ACK 14. TxReqRelINVITE / 103 INVITE 15. Rx SIP/2.0 / 103 INVITE 16. CancelDestroy 17. Rx SIP/2.0 / 103 INVITE 18. CancelDestroy 19. Unhold SIP/2.0 20. TxReq ACK / 103 ACK 21. TxReqRelINVITE / 104 INVITE 22. Rx SIP/2.0 / 104 INVITE 23. CancelDestroy 24. Rx SIP/2.0 / 104 INVITE 25. CancelDestroy 26. Unhold SIP/2.0 27. TxReq ACK / 104 ACK 28. TxReqRelINVITE / 105 INVITE 29. Rx SIP/2.0 / 105 INVITE 30. CancelDestroy 31. Rx SIP/2.0 / 105 INVITE 32. CancelDestroy 33. Unhold SIP/2.0 34. TxReq ACK / 105 ACK 35. TxReqRelINVITE / 106 INVITE 36. Rx SIP/2.0 / 106 INVITE 37. CancelDestroy 38. Rx BYE / 201 BYE 39. TxResp SIP/2.0 / 201 BYE To me it looks like the channel should indeed be removed as it is indeed dead.. but it remains in the sip show channels listing.. Is this a bug? Has this been run into before by others? Does anyone have a remedy for this? Is there perhaps a function that needs to audit periodically the sip channels list to expunge dead channels that should have been removed long ago but have not? begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Audiocodes MP108
We use the mp-108 fxs units a lot.. also use mp-2000 units for pri_cpe end. Probably the closest thing to your situation is our use of the mp2000 terminating a pri at the z end and sending calls on to asterisk. While it was not without it's flaws I can say that it worked rather well just using ip permitted sip connections.. (no auth or registration as that seemed/worked a little goofy). I know it wasn't exactly what you were looking for but I hope it helps. M. Ehsanul Karim wrote: Does anyone have any success using AudioCodes FXO terminating calls ? Ehsan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SUCCESS - 512 Simultaneous Calls with Digital Recording
In light of the I/O bottleneck problem I'd have to ask why asterisk can't just buffer incoming audio and then flush a complete audio file to disk.. I'm assuming that recordings vary in length.. the problem with this idea is what happens if 50 recordings all complete at the same time.. a dump like that might not be very pretty (a fast drive plus a little scheduler or limiter so that only x number of files get written to disk at a time would probably help out there) but I can imagine that a single file being written is much more efficient and more disk-friendly.. perhaps I don't know what the heck I'm talking about but at least in my mind the theory sounds better than the current 'stream-to-file' method employed by asterisk. Matt Roth wrote: All, This message has generated a lot of responses, so I'm going to address each of them here in an attempt to consolidate the thread. Matt, - I'm very interested in the specifics of your setup. - How much space is on the RAM disk? Currently it is 10 GB. We are upgrading it to 16 GB. - What kind of RAM drive is it? The current RAM is 12 GB DDR2 400 MHz (12 x 1GB) Single Ranked DIMMs. The details for each 1 GB DIMM can be seen here: http://www.samsung.com/Products/Semiconductor/DRAM/DDR2SDRAM/DDR2SDRAMmodule/RegisteredDIMM/M393T2950CZ3/M393T2950CZ3.htm The upgrade will involve adding 2 GB DIMMs to the system, but I don't have the details on these yet. The RAM disk is setup by adding the following kernel command line option to grub.conf: ramdisk_size=10485760 We are running Fedora Core 3 with the most up to date 2.6 SMP kernel. By default the RAM disk's block size is 1024 bytes, so we are formatting it as an ext2 file system with a block size of 1024 bytes using the following command: mke2fs -b 1024 -m 0 /dev/ram0 The block size can easily be changed from the kernel's view (using the kernel command line option ramdisk_blocksize=) or from mke2fs's view (using the -b argument), so please let me know if I can make an easy optimization here. Finally, the RAM disk is mounted using the command: mount /dev/ram0 /digrec A good RAMDISK howto exists at: http://www.vanemery.com/Linux/Ramdisk/ramdisk.html - What format are you recording to? - What codec are the SIP calls being placed over? We are recording to the PCM format and using the G711 uLaw codec. High voice quality is essential to our application (we are a call center) so we partnered with MCI to configure our network for the required bandwidth and chose the highest quality, zero compression codec. We noload all other codecs in order to avoid transcoding on the switch, so we must record to PCM. Later (on a separate server) the recordings are mixed to GSM which provides a 5 to 1 compression ratio with very little artifacts. - We've run into the Avoided deadlock recording issues several times when trying to do - more than 50 concurrent recordings. Changing the ast_channel_lock loop from 10 to 20 has - helped somewhat reduce the warnings and reduce audio gaps on the recordings, but what is - really needed for more robust recording is a configurable recording buffer that wouldn't - freak out if a 10ms delay occurs. Are you saying that these messages indicate a gap in a digital recording? If so, what is the duration of the gap? If it's comparable to a CD skip, I think we can deal with it until a buffer or another solution is implemented. - Good luck and please keep us updated on your progress Thank you. I'll be keeping the list informed of our progress. Zoa, - I suppose you are the person from the digium forum That was actually my boss's boss. We thank you all the way up and down the line for your suggestion. - The reason i recommended you to use a ramdisk is because i think the - problem with recording to disk is saving 20ms of stream 1, then 20 ms of - stream 2, then 20ms of stream 3 etc etc meaning you write everytime - very small things. (with a lot of seeking). Agreed. This is why we hope that decoupling the copy (memory to disk) from Asterisk itself and, more importantly, Asterisk's real-time handling of the call being recorded will be sufficient. For the record, when recording 512 simultaneous calls to the local disk we saw a peek of about 13,000 blocks written per second. - Our best test results were with: - - - buffering the recordings to a ramdisk, then We're doing that, as per your suggestion. - - on low load (at night) copy the files over the network (easy to shape - the pipe, so that you dont overload anything), This way, the memory - buffer will take care of the 'fragmentation' and not your harddisk. If you'll note the format of the recordings and that we'll be recording up to 200,000 minutes of calls a day, with a little quick math you'll realize that it would take 80 to 100 GBs of
[Asterisk-Users] sip invite question
I receive an invite from a vendor device.. U 2005/09/19 09:00:18.991139 66.185.96.32:5060 - 66.185.96.23:5060 INVITE sip:[EMAIL PROTECTED];userphone SIP/2.0..Via: SIP/2.0/UDP voip.livewirenet.com:5060;branch=z9hG4bK6ee64f86 Max-Forwards: 70 To: 3036284320 sip:[EMAIL PROTECTED];tag=as60bbddbc From: sip:[EMAIL PROTECTED];user=phone;tag=286349056 Call-ID: [EMAIL PROTECTED] CSeq: 104 INVITE Contact: sip:66.185.96.32:5060 Content-Type: application/sdp Content-Length: 101 v=0 o=- 1127142018 1127142018 IN IP4 0.0.0.0 s=- c=IN IP4 0.0.0.0 t=0 0 m=audio 4010 RTP/AVP 0 Then I see with (sip debug on) that asterisk sends an ok to itself.. why does it do this? My speculation is that the above invite is doing something incorrect.. but what? begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] 1-800 number
Just call a milliwatt..? C F wrote: Try your local DMV :) On 8/18/05, Derek Whitten [EMAIL PROTECTED] wrote: try calling comcast.. they are always good for at least 15 minutes of hold 18778242288 qworst(qwest) works too.. 1800244 On Thu, 2005-08-18 at 06:28, Adam Vocks wrote: Just call tech support for a large company. Your always on hold longer than 10 minutes! Adam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christoph Eicke Sent: Thursday, August 18, 2005 8:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] 1-800 number I'm trying to solve my Nikotel problem (see previous post) where the problem is that I get a hangup after 2 minutes, therefore I need some number that doesn't cost anything and gives me some audio for a long time... On Thursday 18 August 2005 15:08, Jonathan k. Creasy wrote: What problem are you trying to solve with this? Just stepping out on a limb but it sounds like you are trying to swat a fly with an F-16. -Jonathan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christoph Eicke Sent: Wednesday, August 17, 2005 4:34 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] 1-800 number Hi! I'm searching for a 1-800 number that simply plays music for a long time (3mins) and no one picks up. I've bothered the ATT lines so far when trying out my SIP-PSTN connection but then always someone answered :-) Anyone have a number? Christoph ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- -BEGIN GEEK CODE BLOCK- Version: 3.1 GCS/GIT d-@ s+:+ a? C+++ BLHIS$ U+++ P+ L+++ !E W+++$ N++ o+ K w-- PS+++ PE@ Y+ PGP++ t 5? X !R tv+ b- DI-- D G e+ h r+++ y --END GEEK CODE BLOCK-- -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.6 (GNU/Linux) iD4DBQBDBJMCzGZD3alCWIARAgWjAJj+ulL8T7ejTvoEcEsVZ4nhLpt0AJ9iqD18 KuoT22RWCqHTLzT3DgLvpA== =lHKS -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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 Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] stale nonce
Just trying to get some resolution on this as to why it would work with asterisk stable but not current. *bump* Matt Hess wrote: I just updated one of my stable asterisk systems to head to test it out.. and I'm receiving a interesting log message now in asterisk.. Aug 2 13:20:56 NOTICE[15382]: chan_sip.c:5617 check_auth: stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' (one line per registration) I'm using an AudioCodes mp108.. it worked fine with the latest stable.. registrations were ok, etc.. but now in head it's borked. verbose = 30 debug = 30 sip debug on.. *CLI -- SIP read from 66.185.98.152:5060: REGISTER sip:voip.livewirenet.com SIP/2.0 Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682215623 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682209279 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER Contact: sip:[EMAIL PROTECTED];user=phone;expires=86400 Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 86400 User-Agent: Audiocodes-Sip-Gateway-MP-108 FXS/v.4.40.227.419 Content-Length: 0 --- (13 headers 0 lines)--- Using latest request as basis request Sending to 66.185.98.152 : 5060 (non-NAT) Transmitting (no NAT) to 66.185.98.152:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682215623 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682209279 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 66.185.98.152:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682215623 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682209279 To: sip:[EMAIL PROTECTED];user=phone;tag=as2dd53782 Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=voip.livewirenet.com, nonce=6b3e6e93 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms repose*CLI -- SIP read from 66.185.98.152:5060: REGISTER sip:voip.livewirenet.com SIP/2.0 Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682244895 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682239573 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER Contact: sip:[EMAIL PROTECTED];user=phone;expires=86400 Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 86400 User-Agent: Audiocodes-Sip-Gateway-MP-108 FXS/v.4.40.227.419 Content-Length: 0 --- (13 headers 0 lines)--- Using latest request as basis request Sending to 66.185.98.152 : 5060 (non-NAT) Transmitting (no NAT) to 66.185.98.152:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682244895 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682239573 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 66.185.98.152:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682244895 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682239573 To: sip:[EMAIL PROTECTED];user=phone;tag=as3e01dec9 Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=voip.livewirenet.com, nonce=512349ce Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms repose*CLI -- SIP read from 66.185.98.152:5060: REGISTER sip:voip.livewirenet.com SIP/2.0 Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682282366 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682277041 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER Contact: sip:[EMAIL PROTECTED];user=phone;expires=86400 Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 86400 User-Agent: Audiocodes-Sip-Gateway-MP-108 FXS/v.4.40.227.419 Content-Length: 0 --- (13 headers 0 lines)--- Using latest request as basis request Sending to 66.185.98.152 : 5060 (non-NAT) Transmitting (no NAT) to 66.185.98.152:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682282366 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682277041 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 66.185.98.152:5060: SIP/2.0 401
[Asterisk-Users] codec question
I'm looking for opinions on g726-32 vs. g711u.. They both have decent audio quality.. and looking at the wiki I get the impression that g726 is like the little brother to g711. Yet, I've run into quite a few sip termination vendors who don't support it. Does anyone on the list actively use g726 for anything and what have those experiences been? The g726 codec for me at least seems to be the poor man's solution for lack of g729 capability. (apply clue stick as necessary) begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] list test - ignore me
Just checking to see if the list server died again.. been a few hours since I sent something to the list and I usually see it by now.. begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] stale nonce
I just updated one of my stable asterisk systems to head to test it out.. and I'm receiving a interesting log message now in asterisk.. Aug 2 13:20:56 NOTICE[15382]: chan_sip.c:5617 check_auth: stale nonce received from 'sip:[EMAIL PROTECTED];user=phone' (one line per registration) I'm using an AudioCodes mp108.. it worked fine with the latest stable.. registrations were ok, etc.. but now in head it's borked. verbose = 30 debug = 30 sip debug on.. *CLI -- SIP read from 66.185.98.152:5060: REGISTER sip:voip.livewirenet.com SIP/2.0 Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682215623 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682209279 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER Contact: sip:[EMAIL PROTECTED];user=phone;expires=86400 Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 86400 User-Agent: Audiocodes-Sip-Gateway-MP-108 FXS/v.4.40.227.419 Content-Length: 0 --- (13 headers 0 lines)--- Using latest request as basis request Sending to 66.185.98.152 : 5060 (non-NAT) Transmitting (no NAT) to 66.185.98.152:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682215623 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682209279 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 66.185.98.152:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682215623 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682209279 To: sip:[EMAIL PROTECTED];user=phone;tag=as2dd53782 Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=voip.livewirenet.com, nonce=6b3e6e93 Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms repose*CLI -- SIP read from 66.185.98.152:5060: REGISTER sip:voip.livewirenet.com SIP/2.0 Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682244895 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682239573 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER Contact: sip:[EMAIL PROTECTED];user=phone;expires=86400 Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 86400 User-Agent: Audiocodes-Sip-Gateway-MP-108 FXS/v.4.40.227.419 Content-Length: 0 --- (13 headers 0 lines)--- Using latest request as basis request Sending to 66.185.98.152 : 5060 (non-NAT) Transmitting (no NAT) to 66.185.98.152:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682244895 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682239573 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 66.185.98.152:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682244895 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682239573 To: sip:[EMAIL PROTECTED];user=phone;tag=as3e01dec9 Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] WWW-Authenticate: Digest realm=voip.livewirenet.com, nonce=512349ce Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms repose*CLI -- SIP read from 66.185.98.152:5060: REGISTER sip:voip.livewirenet.com SIP/2.0 Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682282366 Max-Forwards: 70 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682277041 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER Contact: sip:[EMAIL PROTECTED];user=phone;expires=86400 Supported: em,timer,replaces,path Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE Expires: 86400 User-Agent: Audiocodes-Sip-Gateway-MP-108 FXS/v.4.40.227.419 Content-Length: 0 --- (13 headers 0 lines)--- Using latest request as basis request Sending to 66.185.98.152 : 5060 (non-NAT) Transmitting (no NAT) to 66.185.98.152:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682282366 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682277041 To: sip:[EMAIL PROTECTED];user=phone Call-ID: [EMAIL PROTECTED] CSeq: 11 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (no NAT) to 66.185.98.152:5060: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 66.185.98.152;branch=z9hG4bKac1682282366 From: sip:[EMAIL PROTECTED];user=phone;tag=1c1682277041 To:
[Asterisk-Users] test message - ignore me
Haven't seen email since the 29th.. just testing. begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWire Networks adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Sr. Network Engineer tel;work:303-458-5667 x 106 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] account code missing in csv cdr
I should also expand this to note that I have also tested the amaflags
variable and it also has no effect on my sip peer entries' cdr records.
(latest stable asterisk has been tested)
Matt Hess wrote:
My cdrs are missing accountcodes for incoming calls from other
asterisk servers..
I've seen a few people mentioning this on the list and the solution
seems to be setting up a dialplan for incoming calls from a particular
sip peer.. in my opinion this does not scale well at all and I am
looking for a solution to correct this problem.
example sip peer:
[asterisk_gw]
type=friend
accountcode=asteriskgw
host=x.x.x.x
restrictcid=no
fromdomain=voip.livewirenet.com
context=lwn
canreinvite=yes
Incoming calls from the peer lack an accountcode field.. a
NoOp(${ACCOUNTCODE}) is blank for calls coming from the above peer..
forcing an accountcode does indeed work with SetAccount() however, as
I stated, this does not scale.
I should also note that both incoming and outgoing calls involving the
peer (and many others) do not have an account code.. I am not
concerned with outgoing calls to the peer in this case.. just incoming
from the peer..
I also saw in one list post that supposedly the accountcode entry in
the sip peer should not be relied upon.. but no detail or explanation
was given as to why that was..
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[Asterisk-Users] tiny audio drops (blips)
We are receiving multiple audio drop outs on calls .. I've done quite a bit of troubleshooting and it only involves calls that require the Dial(SIP/xxx,,t) for transfers.. as long as the media path goes through the server the audio blips happen.. using ulaw codec, btw. I have been able to align the blips in audio to a specific point involving asterisk.. it seems to happen right at about the time asterisk is dealing with another call.. ie: -- Called [EMAIL PROTECTED] It's really an aggravating thing.. what I am asking is this.. we use sip info for dtmf.. (works great for us).. why must the audio stream be running through asterisk if sip info is being used? The # still goes to the asterisk server.. what is the harm in setting up a fresh call leg and reinviting the media end point (party being transferred) over to the new call? It's not like asterisk needs to or even does receive the dtmf inband if it's using sip info anyway right? A few pointers would be appreciated as to smoothing asterisk out some so that other calls being setup do not affect current calls. begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Intermittent Silence
Almost sounds like you and I have the same problem.. please check out my message (I just sent to the list) and see if it rings any bells would you? Stuart Lester wrote: I am currently experiencing intermittent silences with my asterisk system. The symptoms are as follows: * Both for incoming and outgoing calls, I (and other users) occasionally experience a brief period of silence. * The silence lasts anywhere from 3 to 10 seconds. * It is not due to silence suppression, because the silences generally occur in the middle of sentences. * Silences occur at various times of the day, for calls of various lengths, and at different times during calls. * Sliences have occured for incoming and outgoing calls, and have affected both people on the VoIP side and the PSTN side. * Silences have occured for devices from multiple manufacturers (Currently we are using Polycom IP 300, IP 500, and Grandstream 101). * Affected users are connecting to our asterisk system from multiple geographic areas, using multiple ISPs, and all seem to have reliable and acceptable network connections. * All users are connecting to the asterisk system via SIP. * Multiple carriers have been used, and the issue seems to not be related to any particular carrier. * Call load doesn't appear to be an issue, as the silences have occured when the call in question was the only one on the system. The current asterisk system configuration is as follows: * 1.2 GHz Pentium Celeron 512 GB RAM, Linux 2.4.21 * 10 SIP devices connected * Connected to Carriers via IAX * Using uLaw codec * Using asterisk HEAD, built on June 28th * Running ztdummy * Server is not firewalled. I have tried, to no avail, to find a solution for this issue, and I am having no luck. The issue is perplexing, and quite bothersome to our users. Could anyone suggest any solutions, tests or anything? At this point, I'm getting frantic, and any help would be appreciated. Stuart ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] t.38 support news?
Maybe I've missed it but I'm wondering if there has been any movement towards getting t.38 support into asterisk.. has there been any news? Where is t.38 support at? will it even happen? begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] t.38 support news?
Oh, I was just hoping (more like praying to $DEITY) that it'd be in around July or August.. now in a perfect world I'd love for it to be in stable tomorrow (heck, I'd settle for cvs head) but I know that isn't realistic at all. I was just looking for a warm-fuzzy ray of hope. Nathan C. Smith wrote: The developers have talked about it but there are some core components that need to be modified or rewritten to make t.38 integral to asterisk. The way I understand it: They have it in mind, it is a consideration, but don't hold your breath, it won't show up tomorrow. -Nate -Original Message- From: Matt Hess [mailto:[EMAIL PROTECTED] Sent: Wednesday, March 16, 2005 4:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] t.38 support news? Maybe I've missed it but I'm wondering if there has been any movement towards getting t.38 support into asterisk.. has there been any news? Where is t.38 support at? will it even happen? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] SIP-T Support (I got my head in an SS7 cloud)
I, personally, think a channel driver handling both sigtran and mgcp/megaco would be an ideal setup for bridging the gap between ip and pstn.. especially with the current hardware devices on the market.. but all of that is just opinion.. Kevin P. Fleming wrote: [EMAIL PROTECTED] wrote: But I would like asterisk to accept the IMTs.. and the only way to do that is to find out what DS0 to take and place the calls on based on the SS7 messages. I guess one of the things I'm not clear on is going from SS7 to SIP-T, I'm not sure where the state machine exists.. If that's the case, then you external box is just an SS7-SIP-T translator, right? It's not involved in the media path at all. It seems to me that if Asterisk is going to be involved in setup/teardown of the DS0s, then it needs to be involved in the signaling as well. Probably the best way to achieve this is going to be for Asterisk to support SIGTRAN (or one of the other SS7-over-IP solutions) and use an external SS7-to-SIGTRAN translator (or get SIGTRAN from your telco(s), if they support it). ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] app_conference compile?
Has anybody compiled app_conference as of late? I've already asked on the app_conference devel list but as I'm rather in a hurry my thinking is somebody here has both run into and found a way to get this compiled and running. Using stable asterisk and the most recent app_conference from it's cvs on sourceforge.. begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Multiple gateways for same dial pattern
Just a thought I had on this.. Why not setup a sip peer entry in
sip.conf with a qualify statement in it and send the call to the peer
entry? That way asterisk will know if the peer is alive or not and I
would think it would skip that particular peer accordingly.. that or
it's really late and I've misunderstood what qualify is for.. but it
seems to want to work how I imagine it would.. and from a small test I
just tried it appears to do just that.. but who knows..
Adi Linden wrote:
I have a similar problem. I asked the same question in a message to the
list a few days ago titles IAX outgoing redundancy. It would seem
app_dial would need to have some code added to it to have two different
kind of timeouts, one an answer timeout (which is the current timeout in
the Dial() command) and the other a ringing timeout (which would be a
timeout to confirm the call has been passed on to the PSTN
successfully).
I don't know if ringing timeout is how I would describe it. More like a
network timeout. I can describe another scenario, based on what I
originally posted:
exten = _9737,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
exten = _9737,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
If I physically pull the netowrk plug on 172.17.99.5 Asterisk will just
sit there for about a minute before continuing to 172.17.99.6. And the
same scenario holds true for IAX instead of SIP.
Adi
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Re: [Asterisk-Users] Cisco Phones
I may be going out on a limb here but maybe not everyone is on both the users and biz lists as not everyone wants commercial adverts in their email. Note the name of the list.. Asterisk Users Mailing List - Non-Commercial Discussion sorta says it all right there.. Garrett Smith wrote: Matt: I posted this to the right list. I post great deals to this list as not everyone is a member of both lists. Garrett -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Monday, January 03, 2005 5:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco Phones Garrett Smith wrote: I have about (500) more of those NEW IN BOX CP-7960's that many of you purchased last month. If anyone needs more please let me know. You have posted to the wrong list. This type of mail should be directed to asterisk-biz begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Cisco Phones
Talk about a seriously lame excuse.. if I want to save money on the products you sell.. I'll seek you out or lurk on the biz list.. there is no excuse for spamming the thousands on this mailing list to sell to a few hundred of them.. such arrogance makes me sick.. get your own mail list of customers with whom you have done business with previously and work off that.. I suggest those that do buy from this vendor.. stop.. and find an alternative, community friendly, internet etiquette aware vendor. Garrett Smith wrote: I wouldn't consider it an advertisement. There was no price, etc. I was simply telling those that ordered from me previously I have more. It is easier to send one email to 100's, then 100's of emails. If you do not want to save money, nor want the rest of your fellow members to have the opportunity to, I can stop. Fortunately, there is enough interest in these phones coming from the non-business discussion list to warrant a small email alerting everyone to a great deal. I believe great pricing from supplier has been brought to everyone's attention numerous times in the past... Garrett -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Monday, January 03, 2005 7:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco Phones I may be going out on a limb here but maybe not everyone is on both the users and biz lists as not everyone wants commercial adverts in their email. Note the name of the list.. Asterisk Users Mailing List - Non-Commercial Discussion sorta says it all right there.. Garrett Smith wrote: Matt: I posted this to the right list. I post great deals to this list as not everyone is a member of both lists. Garrett -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Riddell Sent: Monday, January 03, 2005 5:39 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Cisco Phones Garrett Smith wrote: I have about (500) more of those NEW IN BOX CP-7960's that many of you purchased last month. If anyone needs more please let me know. You have posted to the wrong list. This type of mail should be directed to asterisk-biz ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] upgraded source now ata's ring but stop silence on inbound calls
I reported this on dev yesterday.. I thought I saw it fixed in dev but not stable according to the cvs list.. Modified Files: chan_sip.c Log Message: Minor ACk fix (bug #2687, again) So the stable version is still borked.. but head should be cleared up..heh, stable ain't that stable right now ;) John Hill wrote: I was doing a daily make update for asterisk. On the 19th the new version compiled fine. I installed it. All of my ata 186's can call out to pstn etc. All inbound calls, the phones ring but when you pickup, just silence both local and remote with no complaints in the cli. I backed down to the r 1.0 1.0.3 on the 20th CVS-v1-0-12/20/04-18:24:52 it worked ok. Yesterday I did a cvs update on the 1.0 source code, recompiled and now it does the same thing as the dev source. Calls go out no problem but inbound rings but are dead upon answer. I had to return to the CVS-v1-0-12/20/04-18:24:52 code to get it to work again. Have I missed a configuration change somewhere? Thanks John Hill ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] upgraded source now ata's ring but stop silenceon inbound calls
Fyi, just saw a fix go through for stable :) John Hill wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Tuesday, December 21, 2004 10:21 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] upgraded source now ata's ring but stop silenceon inbound calls I reported this on dev yesterday.. I thought I saw it fixed in dev but not stable according to the cvs list.. Modified Files: chan_sip.c Log Message: Minor ACk fix (bug #2687, again) So the stable version is still borked.. but head should be cleared up..heh, stable ain't that stable right now ;) John Hill wrote: I was doing a daily make update for asterisk. On the 19th the new version compiled fine. I installed it. All of my ata 186's can call out to pstn etc. All inbound calls, the phones ring but when you pickup, just silence both local and remote with no complaints in the cli. I backed down to the r 1.0 1.0.3 on the 20th CVS-v1-0-12/20/04-18:24:52 it worked ok. Yesterday I did a cvs update on the 1.0 source code, recompiled and now it does the same thing as the dev source. Calls go out no problem but inbound rings but are dead upon answer. I had to return to the CVS-v1-0-12/20/04-18:24:52 code to get it to work again. Have I missed a configuration change somewhere? Thanks John Hill I downloaded the tar release 1.0.3 and it works. The stable cvs yesterday failed. I have not tried the cvs dev or stable. I'll just sit on the release. --john ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Lets try this again then! Q: SIP error from dialplan I suspect!
is this current cvs or something? It looks completely abnormal for stable.. seems you are doing a lot of extra stuff you don't need to.. I'd see if just this works for you.. exten = 800,1,Dial(SIP/800,60) exten = 800,2,VoiceMail(800) also.. why disallow all and then allow most everything? seems like you are trying to over think things.. no offense. why not slim down your peer entry a bit? ie: [800] type=friend username=800 secret=password callerid=800 host=dynamic dtmfmode=inband mailbox=800 nat=yes canreinvite=no David Uzzell wrote: I am playing with the dialplan to get it working and I have a challange with this error. I can't find what it means on the wiki :( Any sugestions would be helpful at being able to forward it to the SIP phone if it is online and avaliable but then let that fail and drop into voicemail if it is not online or is busy. cheers David -- Executing Dial(IAX2/[EMAIL PROTECTED]/3, SIP/800|5) in new stack Dec 21 00:15:57 NOTICE[3922]: app_dial.c:800 dial_exec: Unable to create channel of type 'SIP' (cause 3) == Everyone is busy/congested at this time -- Executing WaitExten(IAX2/[EMAIL PROTECTED]/3, ) in new stack The Extensions.conf file for that section is exten = s,1,Wait,1 exten = s,n,Answer exten = s,n,DigitTimeout,3 exten = s,n,ResponseTimeout,5 exten = s,n,Dial(SIP/800,5) exten = s,n,Waitexten exten = s,n,Playback,voicemail/default/801/unavail exten = s,n,Voicemail,801 exten = s,n,Goto,t|1 and I have in sip.conf [800] type=friend regexten=800 username=800 secret=password callerid=800 host=dynamic ;dtmfmode=inband mailbox=800 nat=yes canreinvite=no qualify=yes disallow=all allow=gsm allow=speex allow=ilbc allow=ulaw allow=alaw ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] voicemail cuts off / hangs up
Is an audio card a prerequisite for using voicemail? I never saw that it was and such a requirement would seem absurd but I thought I'd at least ask. Michael K. Rodriguez User wrote: If I am not mistaken, I believe the dial command is omitted if you do not have a sound card configured on your system (loaded module). -michael On 12/2/04 1:07 AM, Matt Hess [EMAIL PROTECTED] wrote: Does cvs tag v1-0 not have a dial command? I do not seem to have one.. dial No such command 'dial' (type 'help' for help) Henry Devito wrote: Ok try this Login into console Set verbose 15 Dial (extension of VoiceMailMain app) Dial mailbox number Dial password Hangup Does it still die? See my example below asterisk*CLI dial 777 -- Executing VoiceMailMain(OSS/dsp, ) in new stack Console call has been answered -- Playing 'vm-login' (language 'en') asterisk*CLI dial 500 -- Playing 'vm-password' (language 'en') asterisk*CLI dial 1234 -- Playing 'vm-youhave' (language 'en') -- Playing 'digits/8' (language 'en') -- Playing 'vm-INBOX' (language 'en') -- Playing 'vm-messages' (language 'en') -- Playing 'vm-onefor' (language 'en') -- Playing 'vm-INBOX' (language 'en') -- Playing 'vm-messages' (language 'en') asterisk*CLI hangup -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voicemail cuts off / hangs up Test completed successfully.. test dialplan: exten = 555,1,Answer exten = 555,2,Wait(2) exten = 555,3,Playback(digits/0) exten = 555,4,Playback(digits/1) exten = 555,5,Playback(digits/2) exten = 555,6,Playback(digits/3) exten = 555,7,Playback(digits/4) exten = 555,8,Playback(digits/5) exten = 555,9,Playback(digits/6) exten = 555,10,Playback(digits/7) exten = 555,11,Playback(digits/8) exten = 555,12,Playback(digits/9) exten = 555,13,Busy log: -- Executing Answer(SIP/3036284315-31b3, ) in new stack -- Executing Wait(SIP/3036284315-31b3, 2) in new stack -- Executing Playback(SIP/3036284315-31b3, digits/0) in new stack -- Playing 'digits/0' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/1) in new stack -- Playing 'digits/1' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/2) in new stack -- Playing 'digits/2' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/3) in new stack -- Playing 'digits/3' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/4) in new stack -- Playing 'digits/4' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/5) in new stack -- Playing 'digits/5' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/6) in new stack -- Playing 'digits/6' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/7) in new stack -- Playing 'digits/7' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/8) in new stack -- Playing 'digits/8' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/9) in new stack -- Playing 'digits/9' (language 'en') -- Executing Busy(SIP/3036284315-31b3, ) in new stack == Spawn extension (lwn, 555, 13) exited non-zero on 'SIP/3036284315-31b3' Henry Devito wrote: Try to play a number sound file by using the Playback application, I think the voicemail uses the same app to play the digits. See if that works. exten = 500,1,Playback(digits/3) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voicemail cuts off / hangs up yup.. that's something I thought of as well.. and it's all there.. funny thing is.. I can start asterisk.. login just fine to voice mail.. I try again right away and I get that error that I had sent earlier and get cutoff.. Henry Devito wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 11:47 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] voicemail cuts off / hangs up I'm having a problem with voicemail where the system will allow me to login to the vm box no problem but when it starts tell tell me the number of messages I have it hangs up.. I get you have and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. [*] Ok Did you check to see if the /var/lib/asterisk/sounds/digits/ directory is intact? I had an install at an earlier date from the CVS that did not download all of the sounds. Just a thought
Re: [Asterisk-Users] voicemail cuts off / hangs up
That's what I thought.. so that leaves me still at square one for isolating this issue.. Andrew Kohlsmith wrote: On December 2, 2004 12:12 pm, Matt Hess wrote: Is an audio card a prerequisite for using voicemail? I never saw that it was and such a requirement would seem absurd but I thought I'd at least ask. No, a sound card is not required for voicemail. -A. ___ 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 begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] voicemail cuts off / hangs up
Let me ask this.. does anyone have a suggestion for how I can track down which file or directory asterisk is requesting to open? Dec 1 12:17:11 WARNING[19092]: file.c:1058 ast_waitstream_full: Wait failed (No such file or directory) Matt Hess wrote: That's what I thought.. so that leaves me still at square one for isolating this issue.. Andrew Kohlsmith wrote: On December 2, 2004 12:12 pm, Matt Hess wrote: Is an audio card a prerequisite for using voicemail? I never saw that it was and such a requirement would seem absurd but I thought I'd at least ask. No, a sound card is not required for voicemail. -A. ___ 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 begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] voicemail cuts off / hangs up
I'm having a problem with voicemail where the system will allow me to login to the vm box no problem but when it starts tell tell me the number of messages I have it hangs up.. I get you have and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] voicemail cuts off / hangs up
Well I tried tarring up the directory .. deleting and restoring.. no effect.. verbose shows: -- Playing 'vm-login' (language 'en') -- Playing 'vm-password' (language 'en') -- Playing 'vm-youhave' (language 'en') -- Playing 'digits/3' (language 'en') Dec 1 12:17:11 WARNING[19092]: file.c:1058 ast_waitstream_full: Wait failed (No such file or directory) == Spawn extension (lwn, 3036284307, 1) exited non-zero on 'SIP/voip.livewirenet.com-3c96e000' Chad Scott wrote: Maybe a corrupted voicemail directory? Or maybe the files are numbered incorrectly? Put the system into verbose mode and see what happens on the console when you call it... that should help diagnose the problem. On Dec 1, 2004, at 9:47 AM, Matt Hess wrote: I'm having a problem with voicemail where the system will allow me to login to the vm box no problem but when it starts tell tell me the number of messages I have it hangs up.. I get you have and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. mhess.vcf___ 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 begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] voicemail cuts off / hangs up
I did some testing on this .. created a new vm box.. let vm record a message to it's proper directory.. everything appears fine but again with the login hangup / cutoff problem.. verified user/group permissions on the files created in the directory.. the asterisk server runs as the proper user to be able to access these files but the problem persists.. I went into the inbox folder and did a chmod (detailed below) and then it works fine.. wtf! # ls -al total 6 drwx-- 3 root wheel 512 Oct 1 10:34 . drwxr-xr-x 15 root wheel 512 Dec 1 10:40 .. drwx-- 2 root wheel 512 Dec 1 12:45 INBOX # cd INBOX # ls -al total 110 drwx-- 2 root wheel512 Dec 1 12:45 . drwx-- 3 root wheel512 Oct 1 10:34 .. -rwx-- 1 root wheel 8770 Dec 1 12:45 msg.WAV -rwx-- 1 root wheel 8844 Dec 1 12:45 msg.gsm -rw-r--r-- 1 root wheel256 Dec 1 12:45 msg.txt -rwx-- 1 root wheel 85804 Dec 1 12:45 msg.wav # chmod 770 msg.wav msg.WAV msg.gsm # ls -al total 110 drwx-- 2 root wheel512 Dec 1 12:45 . drwx-- 3 root wheel512 Oct 1 10:34 .. -rwxrwx--- 1 root wheel 8770 Dec 1 12:45 msg.WAV -rwxrwx--- 1 root wheel 8844 Dec 1 12:45 msg.gsm -rw-r--r-- 1 root wheel256 Dec 1 12:45 msg.txt -rwxrwx--- 1 root wheel 85804 Dec 1 12:45 msg.wav # ps.. I'm well aware that running as root is naughty.. it's just temporary.. as I'm trying to get this figured out.. Matt Hess wrote: Well I tried tarring up the directory .. deleting and restoring.. no effect.. verbose shows: -- Playing 'vm-login' (language 'en') -- Playing 'vm-password' (language 'en') -- Playing 'vm-youhave' (language 'en') -- Playing 'digits/3' (language 'en') Dec 1 12:17:11 WARNING[19092]: file.c:1058 ast_waitstream_full: Wait failed (No such file or directory) == Spawn extension (lwn, 3036284307, 1) exited non-zero on 'SIP/voip.livewirenet.com-3c96e000' Chad Scott wrote: Maybe a corrupted voicemail directory? Or maybe the files are numbered incorrectly? Put the system into verbose mode and see what happens on the console when you call it... that should help diagnose the problem. On Dec 1, 2004, at 9:47 AM, Matt Hess wrote: I'm having a problem with voicemail where the system will allow me to login to the vm box no problem but when it starts tell tell me the number of messages I have it hangs up.. I get you have and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. mhess.vcf___ 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 ___ 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 begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] voicemail cuts off / hangs up
it went ahead and made a liar out of me and proceeded to not work the second time around.. and really every time since then.. Matt Hess wrote: I did some testing on this .. created a new vm box.. let vm record a message to it's proper directory.. everything appears fine but again with the login hangup / cutoff problem.. verified user/group permissions on the files created in the directory.. the asterisk server runs as the proper user to be able to access these files but the problem persists.. I went into the inbox folder and did a chmod (detailed below) and then it works fine.. wtf! # ls -al total 6 drwx-- 3 root wheel 512 Oct 1 10:34 . drwxr-xr-x 15 root wheel 512 Dec 1 10:40 .. drwx-- 2 root wheel 512 Dec 1 12:45 INBOX # cd INBOX # ls -al total 110 drwx-- 2 root wheel512 Dec 1 12:45 . drwx-- 3 root wheel512 Oct 1 10:34 .. -rwx-- 1 root wheel 8770 Dec 1 12:45 msg.WAV -rwx-- 1 root wheel 8844 Dec 1 12:45 msg.gsm -rw-r--r-- 1 root wheel256 Dec 1 12:45 msg.txt -rwx-- 1 root wheel 85804 Dec 1 12:45 msg.wav # chmod 770 msg.wav msg.WAV msg.gsm # ls -al total 110 drwx-- 2 root wheel512 Dec 1 12:45 . drwx-- 3 root wheel512 Oct 1 10:34 .. -rwxrwx--- 1 root wheel 8770 Dec 1 12:45 msg.WAV -rwxrwx--- 1 root wheel 8844 Dec 1 12:45 msg.gsm -rw-r--r-- 1 root wheel256 Dec 1 12:45 msg.txt -rwxrwx--- 1 root wheel 85804 Dec 1 12:45 msg.wav # ps.. I'm well aware that running as root is naughty.. it's just temporary.. as I'm trying to get this figured out.. Matt Hess wrote: Well I tried tarring up the directory .. deleting and restoring.. no effect.. verbose shows: -- Playing 'vm-login' (language 'en') -- Playing 'vm-password' (language 'en') -- Playing 'vm-youhave' (language 'en') -- Playing 'digits/3' (language 'en') Dec 1 12:17:11 WARNING[19092]: file.c:1058 ast_waitstream_full: Wait failed (No such file or directory) == Spawn extension (lwn, 3036284307, 1) exited non-zero on 'SIP/voip.livewirenet.com-3c96e000' Chad Scott wrote: Maybe a corrupted voicemail directory? Or maybe the files are numbered incorrectly? Put the system into verbose mode and see what happens on the console when you call it... that should help diagnose the problem. On Dec 1, 2004, at 9:47 AM, Matt Hess wrote: I'm having a problem with voicemail where the system will allow me to login to the vm box no problem but when it starts tell tell me the number of messages I have it hangs up.. I get you have and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. mhess.vcf___ 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 ___ 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 begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] voicemail cuts off / hangs up
yup.. that's something I thought of as well.. and it's all there.. funny thing is.. I can start asterisk.. login just fine to voice mail.. I try again right away and I get that error that I had sent earlier and get cutoff.. Henry Devito wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 11:47 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] voicemail cuts off / hangs up I'm having a problem with voicemail where the system will allow me to login to the vm box no problem but when it starts tell tell me the number of messages I have it hangs up.. I get you have and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. [*] Ok Did you check to see if the /var/lib/asterisk/sounds/digits/ directory is intact? I had an install at an earlier date from the CVS that did not download all of the sounds. Just a thought. ___ 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 begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] voicemail cuts off / hangs up
Test completed successfully.. test dialplan: exten = 555,1,Answer exten = 555,2,Wait(2) exten = 555,3,Playback(digits/0) exten = 555,4,Playback(digits/1) exten = 555,5,Playback(digits/2) exten = 555,6,Playback(digits/3) exten = 555,7,Playback(digits/4) exten = 555,8,Playback(digits/5) exten = 555,9,Playback(digits/6) exten = 555,10,Playback(digits/7) exten = 555,11,Playback(digits/8) exten = 555,12,Playback(digits/9) exten = 555,13,Busy log: -- Executing Answer(SIP/3036284315-31b3, ) in new stack -- Executing Wait(SIP/3036284315-31b3, 2) in new stack -- Executing Playback(SIP/3036284315-31b3, digits/0) in new stack -- Playing 'digits/0' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/1) in new stack -- Playing 'digits/1' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/2) in new stack -- Playing 'digits/2' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/3) in new stack -- Playing 'digits/3' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/4) in new stack -- Playing 'digits/4' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/5) in new stack -- Playing 'digits/5' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/6) in new stack -- Playing 'digits/6' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/7) in new stack -- Playing 'digits/7' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/8) in new stack -- Playing 'digits/8' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/9) in new stack -- Playing 'digits/9' (language 'en') -- Executing Busy(SIP/3036284315-31b3, ) in new stack == Spawn extension (lwn, 555, 13) exited non-zero on 'SIP/3036284315-31b3' Henry Devito wrote: Try to play a number sound file by using the Playback application, I think the voicemail uses the same app to play the digits. See if that works. exten = 500,1,Playback(digits/3) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voicemail cuts off / hangs up yup.. that's something I thought of as well.. and it's all there.. funny thing is.. I can start asterisk.. login just fine to voice mail.. I try again right away and I get that error that I had sent earlier and get cutoff.. Henry Devito wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 11:47 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] voicemail cuts off / hangs up I'm having a problem with voicemail where the system will allow me to login to the vm box no problem but when it starts tell tell me the number of messages I have it hangs up.. I get you have and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. [*] Ok Did you check to see if the /var/lib/asterisk/sounds/digits/ directory is intact? I had an install at an earlier date from the CVS that did not download all of the sounds. Just a thought. ___ 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 begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] voicemail cuts off / hangs up
Does cvs tag v1-0 not have a dial command? I do not seem to have one.. dial No such command 'dial' (type 'help' for help) Henry Devito wrote: Ok try this Login into console Set verbose 15 Dial (extension of VoiceMailMain app) Dial mailbox number Dial password Hangup Does it still die? See my example below asterisk*CLI dial 777 -- Executing VoiceMailMain(OSS/dsp, ) in new stack Console call has been answered -- Playing 'vm-login' (language 'en') asterisk*CLI dial 500 -- Playing 'vm-password' (language 'en') asterisk*CLI dial 1234 -- Playing 'vm-youhave' (language 'en') -- Playing 'digits/8' (language 'en') -- Playing 'vm-INBOX' (language 'en') -- Playing 'vm-messages' (language 'en') -- Playing 'vm-onefor' (language 'en') -- Playing 'vm-INBOX' (language 'en') -- Playing 'vm-messages' (language 'en') asterisk*CLI hangup -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 4:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voicemail cuts off / hangs up Test completed successfully.. test dialplan: exten = 555,1,Answer exten = 555,2,Wait(2) exten = 555,3,Playback(digits/0) exten = 555,4,Playback(digits/1) exten = 555,5,Playback(digits/2) exten = 555,6,Playback(digits/3) exten = 555,7,Playback(digits/4) exten = 555,8,Playback(digits/5) exten = 555,9,Playback(digits/6) exten = 555,10,Playback(digits/7) exten = 555,11,Playback(digits/8) exten = 555,12,Playback(digits/9) exten = 555,13,Busy log: -- Executing Answer(SIP/3036284315-31b3, ) in new stack -- Executing Wait(SIP/3036284315-31b3, 2) in new stack -- Executing Playback(SIP/3036284315-31b3, digits/0) in new stack -- Playing 'digits/0' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/1) in new stack -- Playing 'digits/1' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/2) in new stack -- Playing 'digits/2' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/3) in new stack -- Playing 'digits/3' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/4) in new stack -- Playing 'digits/4' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/5) in new stack -- Playing 'digits/5' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/6) in new stack -- Playing 'digits/6' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/7) in new stack -- Playing 'digits/7' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/8) in new stack -- Playing 'digits/8' (language 'en') -- Executing Playback(SIP/3036284315-31b3, digits/9) in new stack -- Playing 'digits/9' (language 'en') -- Executing Busy(SIP/3036284315-31b3, ) in new stack == Spawn extension (lwn, 555, 13) exited non-zero on 'SIP/3036284315-31b3' Henry Devito wrote: Try to play a number sound file by using the Playback application, I think the voicemail uses the same app to play the digits. See if that works. exten = 500,1,Playback(digits/3) -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 3:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voicemail cuts off / hangs up yup.. that's something I thought of as well.. and it's all there.. funny thing is.. I can start asterisk.. login just fine to voice mail.. I try again right away and I get that error that I had sent earlier and get cutoff.. Henry Devito wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Matt Hess Sent: Wednesday, December 01, 2004 11:47 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] voicemail cuts off / hangs up I'm having a problem with voicemail where the system will allow me to login to the vm box no problem but when it starts tell tell me the number of messages I have it hangs up.. I get you have and it dies right there.. I'm running cvs tag v1-0.. what might be causing this? I looked through my mail list archive and didn't notice anything like this.. [*] Ok Did you check to see if the /var/lib/asterisk/sounds/digits/ directory is intact? I had an install at an earlier date from the CVS that did not download all of the sounds. Just a thought. ___ 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
[Asterisk-Users] OT: anyone using pointone?
Sorry for the OT message but I'm very curious to see if anyone on this list uses pointone for long distance sip call termination? We've been having an off and on problem with them saying they do not support sip message with a fqdn in the from field.. which to me appears to be a breakage of the sip rfc.. and to top it off all our other calls process through them just fine expect to a current problem area code out in California.. I feel they are giving us a very generic white-washed answer and do not wish to actually provide good customer service.. opinions, comments, or cuss words? begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] cheap gig switch? smc, netgear, or 3com?
There was a thread on NANOG a while back about dell switches and the opinion at the time seemed almost in complete agreement - dell switches stink for everything but pure ipv4 shuffle packets.. unmanaged without any features. They are not ciscos at all.. they have a cisco like interface but then again so does zebra.. but that doesn't make it a cisco either. And imho, being the 'wal-mart' of something isn't necessarily a good thing.. even wal-mart sells some total junk (to put it lightly). Brian Roy wrote: Bleh, what group said this? The 3Com group? Dell is the WalMart of the hardware world. Their pricing is better because they build efficiencies. I have 33XX 54XX and I just bought my first 6024 Layer 3 QOS ready switch. These things are nothing but Ciscos in sheep's clothing. They have been rock solid for me and others that I know who use them. -Chuji On Wed, 20 Oct 2004 14:47:21 -0600, Matt Hess [EMAIL PROTECTED] wrote: Remember, you pay for what you get.. especially with Dell networking equipment. I have heard about several groups who tried the dell switches only to give up on them because the dell switches just didn't perform. Yes, price-wise they look good.. but as far as performance goes.. (that is assuming you want high/solid performance) I'd look elsewhere. ___ 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 begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] sip call echo cancellation
I haven't found much info that stands out to me in regards to echo cancellation in sip to sip calls.. My setup is this.. I have a key system connection to an audiocodes mp-108 which connects to asterisk via sip and asterisk passes the call over to a ser proxy and ser passes the call to either an internet voice terminator for long distance or a local pstn gateway.. the local gateway is a lucent max tnt with 10.1.1 software load.. it dumps the sip calls onto pri trunks.. I'm still getting feedback from the office as to whether or not echo is heard on local or long distance.. incoming outgoing etc.. so I know which part to troubleshoot.. if the echo is on egress or ingress.. I know it isn't the local key system with echo as it's almost all digital apart from the analog connection from key system to audiocodes sip gateway as I also have a few ip phones which connect to asterisk that experience similar echo issues.. remote callers don't seem to hear any echo so it would appear to be an impedance issue somewhere remotely on the pstn.. My thinking is that the mismatch may be on the local tnt gateway but I'm not 100% positive of that.. I know ingress calls to the sip network have echo and those flow through the pri lines to the tnt so I'm starting there with that gateway but I'm looking for something possibly on asterisk that can help with pure sip call echo cancellation.. does anyone have any ideas as to what may help? begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] sip call echo cancellation
I have echo cancellation on in the mp-108 as well as silence suppression.. tinkering with these two settings has not yet yielded any happy results.. which is why I'm asking if the asterisk system has any possible echo cancellation that may help alleviate this issue as it's not only seen on the mp-108 but a few sip phones as well such as a budgetone 2 and an ipdialog.. I have a feeling this has to do with voice output levels.. but it's just a hunch that adding a bit of a reduction to the output audio on the pri trunks will help.. as I remember it.. if point A calls point B and point A hears echo and point B does not then it's something like an impedance problem on the far end.. or wire center (CO) at that point.. but an impedance mismatch can occur also at a pstn ingress point such as the ILEC side of the pri lines.. or maybe I'm wrong.. Steve Clark wrote: Matt Hess wrote: I haven't found much info that stands out to me in regards to echo cancellation in sip to sip calls.. My setup is this.. I have a key system connection to an audiocodes mp-108 which connects to asterisk via sip and asterisk passes the call over to a ser proxy and ser passes the call to either an internet voice terminator for long distance or a local pstn gateway.. the local gateway is a lucent max tnt with 10.1.1 software load.. it dumps the sip calls onto pri trunks.. I'm still getting feedback from the office as to whether or not echo is heard on local or long distance.. incoming outgoing etc.. so I know which part to troubleshoot.. if the echo is on egress or ingress.. I know it isn't the local key system with echo as it's almost all digital apart from the analog connection from key system to audiocodes sip gateway as I also have a few ip phones which connect to asterisk that experience similar echo issues.. remote callers don't seem to hear any echo so it would appear to be an impedance issue somewhere remotely on the pstn.. My thinking is that the mismatch may be on the local tnt gateway but I'm not 100% positive of that.. I know ingress calls to the sip network have echo and those flow through the pri lines to the tnt so I'm starting there with that gateway but I'm looking for something possibly on asterisk that can help with pure sip call echo cancellation.. does anyone have any ideas as to what may help? ___ 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 In the MP-108 try turning on Silence Suppression and Echo Cancellation. begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] SER or not to SER?
I use a failure route in ser for the call to be sent to the voicemail system.. I use ser as mainly a primary router for sip messages that sits in the center of a ring of asterisk servers that feed the clients.. Iqbal wrote: Hi i am stuck with the same dilemma, as the original poster I have setup ser now (with the helpful pointer from Girish..tks mate) and can do Ip --- Ip calls, and IP ---pstn (via cisco box), all via ser, however I also have asterisk installed, and now am wondering where I use asterisk, it was/is suggested I use it for all pbx functions such as voicemail etc, however I cant seem to see how on a call not answered howto get ser to send to asterisk. I also am looking at the prepaid billing option, and hence the main reason for asterisk, but unless all calls flow via asterisk instead of ser I cant see the point of astcc, and if they do all flow via asterisk, then why put ser infront... tks iqbal On 10/21/2004, Darren Sessions [EMAIL PROTECTED] wrote: We use SER + Asterisk. One heck of a powerful combination. On Oct 21, 2004, at 11:59 AM, Nahuel Alejandro Ramos wrote: Hi everyone, I have some doubt about use or not to use SER. I need a solution using a single linux box that manages, aproximatly 500-1000 registred SIP users, but not more than 50 simultaneouly calls. My plan is to use between 3 to 5 Cisco 26xx PSTN gateway in diferents cities of my country (Argentine) connected through Internet (with public IP). I was searching for SER solutions (and works perfectly) but it does not support Prepaid Billing. So I post a message (on SerUsers maillist) and everybody said me to use Asterisk to use a Prepaid Billing App., so I install Asterisk. I googled, read this maillist (and post some message) and I receive some helpful answers recomending me to install ASTCC, so I install it too and work perfectly too. My questions (if someone could help me) are : 1) What platform (hardware) do I need to support my call flow (500-1000 registers and 50 simultaneouly calls)? 2) Do I need to install SER? 3) If YES, do I need to register my SIP clients on SER and forward all the calls to Asterisk? 3) If NO, do I need to register my SIP clients on Asterisk and forward all the calls to SER? 4) Can I use only Asterisk to REGISTER, AUTH, ACC, ROUTE CALLS, etc, SIP clients? 5) Could I use extension.conf file to route my calls to my diferents Cisco PSTN GW? 6) And how can I use MySQL instead of file? (I have created the DB and tables but I do not know how to make Asterisk use it instead the extension.conf file) 7) I found easy to use only Asterisk, but I have read that it uses to much CPU and memory, is that true? 8) Could anyone some me information about how to configure Asterisk to receive calls through Cisco PSTN GW? 9) THANK YOU VERY VERY MUCH!!! Nahuel Ramos. ___ 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 ___ 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 begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] Fax detection in voip channel
Ironically, I just got back to my desk from faxing an 18 page document.. ulaw codec.. call path: sip gateway - asterisk - max tnt - pri lines/pstn I haven't really had many troubles with fax.. apart from when I was using a particular voip provider named voiplist.. they refused to support it saying that and I quote: the service is voice over ip not fax over ip how silly is that.. But I do agree.. a nice open t.38 platform would be wonderful.. inside of asterisk.. on media gateways.. everywhere.. but it isn't there yet.. usedcanon wrote: -Original Message- From: Lee Howard [mailto:[EMAIL PROTECTED] Sent: 21 October 2004 23:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: usedcanon Subject: Re: [Asterisk-Users] Fax detection in voip channel On 2004.10.21 14:49 usedcanon wrote: Hi All, Is it possible to detect an incomming fax just as it is possible with Answer on a Zap channel. If not do others find the possibility of this enhancement useful too? Doing fax over SIP or IAX would be a frustrating effort, and a complete waste of time, IMO. See: http://www.opencall.org/faq/x47.html If you don't believe me, go ahead and actually *try* to send/receive a fax through a WAN/internet VoIP connection. You'll probably get tolerable results with SIP-fax on a LAN, but run it through a VoIP provider over the internet, and you'll have a mess, even if the codec is ULAW/ALAW What you really want is a T.38 channel driver. Lee. I understand what you are saying however there are scenarios where fax over voip works fine, I have tested (briefly) with spandsp and have done so sucessfully. as a sperate solutions we use mediatrix gateways very successfully for fax transmision over IP, our IP network is private and does not touch the internet so we can gaurantee (to some extent) bandwidth and quality. A T.38 solution would be most desirable, no doubt, but is there one ? I don't even see a mention of it anywhere. Umar ___ 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 begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] cheap gig switch? smc, netgear, or 3com?
Remember, you pay for what you get.. especially with Dell networking equipment. I have heard about several groups who tried the dell switches only to give up on them because the dell switches just didn't perform. Yes, price-wise they look good.. but as far as performance goes.. (that is assuming you want high/solid performance) I'd look elsewhere. Jon Radon wrote: Best value in gig switches right now is Dell. Go to Dell Small Business and keep an eye out for some deals. They have a pretty good one going on now for their 2000 series. http://www1.us.dell.com/content/products/compare.aspx/2000_workgroup_gig?c=uscs=04l=ens=bsd *Not affiliated with dell.. their 16 port and 24 port are a great buy. :) On Wed, 20 Oct 2004 08:39:01 -0400, dean collins [EMAIL PROTECTED] wrote: I have one of these, works great but failed about 6 months into it's life, was replaced on the spot (in Australia (I'm originally from there) but you had to drive it to them with the original receipt for the handover). Does anyone know if this is a worldwide warranty? Has anyone in NY tried to claim? Where was it etc? Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jay Wilton Sent: Wednesday, October 20, 2004 4:08 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] cheap gig switch? smc, netgear, or 3com? Hello, The Smc 8508T goes for about $95, jumbo frame support, lifetime warranty but no QOS. The Netgear GS608 is $ 100, no jumbo frames, 1 year warranty, QOS, gig latency 10U max. The 3com switch reviews that I read were not happy. Does anyone hate or love their home switch? I doubt the jumbo frame support would help voip traffic, but it seems like it wouldn't hurt. I was planning on doing the QOS on linux. Gig support is wanted for file transfers and the future. Thanks to all you nice asterisk people and a few of the mean ones. Jay __ Do you Yahoo!? Yahoo! Mail Address AutoComplete - You start. We finish. http://promotions.yahoo.com/new_mail ___ 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 begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] quiet term
Is there any option in Asterisk to create a quiet termination? I'm looking for something similar to the 600 ohm impedance lines that COs used to have to check for return loss. begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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] quiet term
Yes.. just something to /dev/null audio sent by the calling party.. So far what I'm doing is just an Answer() followed by a wait(1800) to give the tester some time before it kills the call.. I figure that way asterisk picks up the call and does absolutely nothing with it.. and it appears to work (so far). Steven Critchfield wrote: On Wed, 2004-10-13 at 11:26 -0600, Matt Hess wrote: Is there any option in Asterisk to create a quiet termination? I'm looking for something similar to the 600 ohm impedance lines that COs used to have to check for return loss. wouldn't that be like the milliwatt app but with out the tone generation? begin:vcard fn:Matt Hess n:Hess;Matt org:LiveWireNet adr;dom:;;4577 Pecos St;Denver;CO;80211 email;internet:[EMAIL PROTECTED] title:Senior Network Engineer tel;work:303-458-5667 tel;fax:303-458-5725 x-mozilla-html:FALSE url:http://www.livewirenet.com/ version:2.1 end:vcard ___ 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
