May 22 23:07:50 WARNING[3119]: Unexpected freqency 44100 I can't believe i didn't see that! i spent ages staring at those damn logs...
i'm sure that will fix it. thanks r > Send Asterisk-Users mailing list submissions to > [email protected] > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body 'help' to > [EMAIL PROTECTED] > > You can reach the person managing the list at > [EMAIL PROTECTED] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Asterisk-Users digest..." > > > Today's Topics: > > 1. Initial second lost on SIP phones (Pieter Claassen) > 2. How to detect call forwarding to voicemail (Nitin Gupta) > 3. I've broken voicemail (Robbie Hughes) > 4. Re: Re: I get MOH when the caller hangs up (Steve Totaro) > 5. Re: How to detect call forwarding to voicemail > (Eric "ManxPower" Wieling) > 6. Re: Office to Office via IAX2 problems ([EMAIL PROTECTED]) > 7. Re: I've broken voicemail (Patrick) > 8. FAX and Asterisk (Rene Nelson) > 9. Re: I've broken voicemail (Mark Phillips) > 10. Re: FAX and Asterisk (Lee Howard) > 11. Re: How to detect call forwarding to voicemail (Leo Ann Boon) > 12. Re: Office to Office via IAX2 problems ([EMAIL PROTECTED]) > 13. Timeframe for QueueStatus values ([EMAIL PROTECTED]) > 14. Re: Timeframe for QueueStatus values (BJ Weschke) > 15. CallerID (Greg Oliver) > 16. Re: Help Avaya 4606 (Tom Lynn) > 17. US telco lingo (Eric Bishop) > 18. RE: US telco lingo (Kerry Garrison) > 19. Faxing - machines stop talking, line stays up (Warrick Zedi) > 20. Re: FAX and Asterisk (Alejandro Vargas) > 21. TDM400P , "ztcfg ?vv error ", "Does it have to do with my PC > hardware ?" (John Joseph) > 22. [Fwd: [Asterisk-Users] Faxing - machines stop talking, line > stays up] (Warrick Zedi) > 23. SIP session number (Giordano Grandis) > 24. Logger rotate & master.csv (Asterisk) > 25. Free/Open pci telco card (Kai Ober) > 26. Re: voicemail access on the Thomson ST2030 ? > (Louis-David Mitterrand) > 27. A call from a call file always does a redial? (Remco Barende) > 28. Re: Deadlocks in 1.2.7.1 (Philipp Ott) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 22 May 2006 23:27:36 +0200 > From: Pieter Claassen <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Initial second lost on SIP phones > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > I find that when asterisk answers the phone, the initial second or so is > lost. > I can imagine that echotraining can do this, but this is between SIP > phones > and I don't think there is any echotraining enabled? > > BTW. Asterisk is definitely playing sounds that first second (The CLI > would > indicate that it would play a beep but I just won't hear it). > > Any comments appreciated. > Pieter > > > ------------------------------ > > Message: 2 > Date: Mon, 22 May 2006 14:31:17 -0700 > From: "Nitin Gupta" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] How to detect call forwarding to voicemail > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > Hi, > Is there anyway in Asterisk to know that outgoing call has been forwarded > to voicemail by the callee system? > > Some of my users don't want to connect the call if its forwarded to callee > voicemail, so I am wondering if theres anyway to identify this in Asterisk > and drop the call. > > Thanks > Nitin > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20060522/c0617a64/attachment-0001.htm > > ------------------------------ > > Message: 3 > Date: Mon, 22 May 2006 23:11:15 +0100 > From: Robbie Hughes <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] I've broken voicemail > To: "[email protected]" > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="US-ASCII" > > I went to put in the new sound files over the weekend, but forgot to > backup > the custom folder and lost my custom digital receptionist files. > I then had to copy the old files back from a duplicate machine. > > The problem is now though that voicemail just hangs up when I dial it. > > Other apps work - *70 for example gives me 'call waiting...activated' so I > know it's accessing the files correctly, and the gsm files themselves are > all chown asterisk, chgrp asterisk chmod 644 * etc so they can be read and > there shouldn't be a permissions issue...but voicemail is not working. > > In asterisk I get the following: > > > asterisk*CLI> set verbose 999 > Verbosity was 0 and is now 999 > -- Executing Answer("SIP/1607-a359", "") in new stack > -- Executing Wait("SIP/1607-a359", "1") in new stack > -- Executing VoiceMailMain("SIP/1607-a359", "default") in new stack > == Spawn extension (from-internal, *98, 3) exited non-zero on > 'SIP/1607-a359' > -- Executing Macro("SIP/1607-a359", "hangupcall") in new stack > -- Executing ResetCDR("SIP/1607-a359", "w") in new stack > -- Executing NoCDR("SIP/1607-a359", "") in new stack > -- Executing Wait("SIP/1607-a359", "5") in new stack > -- Executing Hangup("SIP/1607-a359", "") in new stack > == Spawn extension (macro-hangupcall, s, 4) exited non-zero on > 'SIP/1607-a359' in macro 'hangupcall' > == Spawn extension (from-internal, h, 1) exited non-zero on > 'SIP/1607-a359' > > > And full log gives me > > May 22 23:07:49 DEBUG[3119]: Setting NAT on RTP to 0 > May 22 23:07:49 DEBUG[3119]: Stopping retransmission on > '[EMAIL PROTECTED]' of Response 3706: > Found > May 22 23:07:49 DEBUG[3119]: Setting NAT on RTP to 0 > May 22 23:07:49 DEBUG[3119]: Check for res for 1607 > May 22 23:07:49 DEBUG[3119]: Call from user '1607' is 1 out of 0 > May 22 23:07:49 DEBUG[3119]: build_route: Contact hop: > <sip:[EMAIL PROTECTED]:5060> > May 22 23:07:49 VERBOSE[3119]: -- Executing Answer("SIP/1607-bd04", > "") > in new stack > May 22 23:07:49 VERBOSE[3119]: -- Executing Wait("SIP/1607-bd04", "1") > in new stack > May 22 23:07:49 DEBUG[3119]: Stopping retransmission on > '[EMAIL PROTECTED]' of Response 3707: > Found > May 22 23:07:50 VERBOSE[3119]: -- Executing > VoiceMailMain("SIP/1607-bd04", "default") in new stack > May 22 23:07:50 WARNING[3119]: Unexpected freqency 44100 > May 22 23:07:50 WARNING[3119]: Unable to open fd on > /var/lib/asterisk/sounds/vm-login.wav > May 22 23:07:50 WARNING[3119]: Unable to open vm-login (format ulaw): No > such file or directory > May 22 23:07:50 WARNING[3119]: Couldn't stream login file > May 22 23:07:50 VERBOSE[3119]: == Spawn extension (from-internal, *98, > 3) > exited non-zero on 'SIP/1607-bd04' > May 22 23:07:50 VERBOSE[3119]: -- Executing Macro("SIP/1607-bd04", > "hangupcall") in new stack > May 22 23:07:50 VERBOSE[3119]: -- Executing ResetCDR("SIP/1607-bd04", > "w") in new stack > May 22 23:07:50 DEBUG[3119]: cdr_mysql: inserting a CDR record. > May 22 23:07:50 DEBUG[3119]: cdr_mysql: SQL command as follows: INSERT > INTO > cdr > (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration > ,billsec,disposition,amaflags,accountcode) VALUES ('2006-05-22 > 23:07:49','\"GPPlus\" <1607>','1607','*98','from-internal', > 'SIP/1607-bd04','','ResetCDR','w',1,1,'ANSWERED',3,'') > May 22 23:07:50 VERBOSE[3119]: -- Executing NoCDR("SIP/1607-bd04", "") > in new stack > May 22 23:07:50 WARNING[3119]: CDR on channel 'SIP/1607-bd04' not posted > May 22 23:07:50 WARNING[3119]: CDR on channel 'SIP/1607-bd04' lacks end > May 22 23:07:50 VERBOSE[3119]: -- Executing Wait("SIP/1607-bd04", "5") > in new stack > May 22 23:07:55 VERBOSE[3119]: -- Executing Hangup("SIP/1607-bd04", > "") > in new stack > May 22 23:07:55 VERBOSE[3119]: == Spawn extension (macro-hangupcall, s, > 4) > exited non-zero on 'SIP/1607-bd04' in macro 'hangupcall' > May 22 23:07:55 VERBOSE[3119]: == Spawn extension (from-internal, h, 1) > exited non-zero on 'SIP/1607-bd04' > May 22 23:07:55 DEBUG[3119]: update_user_counter(1607) - decrement inUse > counter > May 22 23:07:55 DEBUG[3119]: Stopping retransmission on > '[EMAIL PROTECTED]' of Request 102: Found > > > > But nothing there is particularly life-threatening...that I can see at > least? > > Does anyone have any ideas of where I could start to look for this > problem? > I can't find any evidence that I've broken anything but for the fact that > voicemail is broken. > > Any help appreciated. > r > > > > > ------------------------------ > > Message: 4 > Date: Mon, 22 May 2006 18:17:19 -0400 > From: Steve Totaro <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Re: I get MOH when the caller hangs up > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Douglas Garstang wrote: >> Do you have a 'g' option in your dial command? That will cause the dial >> plan to continue executing after they hangup.... I think. >> >> >>> -----Original Message----- >>> From: Tony Mountifield [mailto:[EMAIL PROTECTED] >>> Sent: Monday, May 22, 2006 8:15 AM >>> To: [email protected] >>> Subject: [Asterisk-Users] Re: I get MOH when the caller hangs up >>> >>> >>> In article >>> <[EMAIL PROTECTED]>, >>> Steven Totaro <[EMAIL PROTECTED]> wrote: >>> >>>> Exten = h,1,hangup ? >>>> >>> No, there's never any need to call Hangup in the h extension, because >>> by the time h is called, the call is already hung up, by definition. >>> >>> Cheers >>> Tony >>> -- >>> Tony Mountifield >>> > Would it not be logical that the hangup in the h extension would hang up > the local channel? If the local leg of the call was hungup then why MOH? > > > ------------------------------ > > Message: 5 > Date: Mon, 22 May 2006 17:21:00 -0500 > From: "Eric \"ManxPower\" Wieling" <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] How to detect call forwarding to > voicemail > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Nitin Gupta wrote: >> Hi, >> Is there anyway in Asterisk to know that outgoing call has been >> forwarded >> to voicemail by the callee system? >> >> Some of my users don't want to connect the call if its forwarded to >> callee >> voicemail, so I am wondering if theres anyway to identify this in >> Asterisk >> and drop the call. > > Your first question should be "Can the telco inform the calling > equipment that the call has been forwarded to voicemail?" As far as I > know, the answer to that is "no". > > -- > Now accepting new clients in Birmingham, Atlanta, Huntsville, > Chattanooga, and Montgomery. > > > ------------------------------ > > Message: 6 > Date: Mon, 22 May 2006 16:55:12 -0500 > From: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Office to Office via IAX2 problems > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii"; format=flowed > > Thanks, Noah. > > I'm at Office A. I ssh into Office B * box using putty. While > logged on to Office B via putty, I can ssh back into Office A * box > by typing ssh [EMAIL PROTECTED] > > Ping times can be 50-1000ms. I've tried qualify=yes, no qualify > statement at all, qualify=1500, and qualify=2000 none helped. Each > time I make a change, I issue the reload command from the > CLI. Should I use a different command? > > Thanks, > Doug > > At 03:30 PM 5/22/2006, you wrote: >>Hi Doug - >> >>Just to cover all the bases. Can one machine talk to the other at >>all? Can you ssh from one box to another (if you don't use ssh, can >>you telnet to an open tcp port)? If not, it is surely a routing >>issue. >> >>If you can connect via non-asterisk methods, you might try increasing >>your qualify value to something higher (qualify=1500), or just remove >>it altogether for testing. It might be that the latency is high >>enough that the connection consistently fails to qualify. (What are >>the ping times, BTW?) >> >>I'll second Eric's advice to not use a DNS name for the host, even in >>your final setup. >> >>- Noah > > > > > ------------------------------ > > Message: 7 > Date: Tue, 23 May 2006 00:49:36 +0200 > From: Patrick <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] I've broken voicemail > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain > > On Mon, 2006-05-22 at 23:11 +0100, Robbie Hughes wrote: > [snip] >> May 22 23:07:50 WARNING[3119]: Unexpected freqency 44100 > > Aren't Asterisk sound files supposed to use 8KHz? Did you perhaps forgot > to transform the wav(s) you are using now to 8KHz versions? > > Regards, > Patrick > > > > > > ------------------------------ > > Message: 8 > Date: Mon, 22 May 2006 16:51:35 -0600 > From: "Rene Nelson" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] FAX and Asterisk > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > I want to accept faxes via SIP/IAX2 (yes I've read the posts that it isnt > reccomended). My PBX is 100% Virtual with the exception of one IAX > connection to bring in the calls from another * box, I have no phone > hardware. I am interested in doing autodetect fax to email. I have found > all kinds of posts to this or that, but all seem to reference Asterisk > 1.0.xor > 1.1.x I am running 1.2.x. > > Can anyone point me in the right direction to get this solution up and > running? > > Thanks > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20060522/3d59b669/attachment-0001.htm > > ------------------------------ > > Message: 9 > Date: Mon, 22 May 2006 18:52:45 -0400 > From: Mark Phillips <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] I've broken voicemail > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain > > I know what he's done. > > He's installed my Alison Keenan wav files without converting them. Try > downloading the sln files instead. > > BTW, G723 and G729 files going up tomorrow. > > Mark > > On Tue, 2006-05-23 at 00:49 +0200, Patrick wrote: >> On Mon, 2006-05-22 at 23:11 +0100, Robbie Hughes wrote: >> [snip] >> > May 22 23:07:50 WARNING[3119]: Unexpected freqency 44100 >> >> Aren't Asterisk sound files supposed to use 8KHz? Did you perhaps forgot >> to transform the wav(s) you are using now to 8KHz versions? >> >> Regards, >> Patrick >> >> >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> Asterisk-Users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > > > ------------------------------ > > Message: 10 > Date: Mon, 22 May 2006 16:05:22 -0700 > From: Lee Howard <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] FAX and Asterisk > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Rene Nelson wrote: > >> I want to accept faxes via SIP/IAX2 (yes I've read the posts that it >> isnt reccomended). My PBX is 100% Virtual with the exception of one >> IAX connection to bring in the calls from another * box, I have no >> phone hardware. I am interested in doing autodetect fax to email. I >> have found all kinds of posts to this or that, but all seem to >> reference Asterisk 1.0.x or 1.1.x I am running 1.2.x. >> >> Can anyone point me in the right direction to get this solution up and >> running? > > > The right thing to do first would be to contact your SIP/IAX2 provider > and find out if your connection to them will always be jitter-free (and > this is not likely to be the case). If it will be then you can do > faxing to their equipment. If not (and this is probably the case), then > you will be wasting a lot of time faxing that way and you should > probably just get a fax account with an on-line fax service. > > Lee. > > > ------------------------------ > > Message: 11 > Date: Tue, 23 May 2006 07:29:26 +0800 > From: Leo Ann Boon <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] How to detect call forwarding to > voicemail > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Eric "ManxPower" Wieling wrote: > >> >> Your first question should be "Can the telco inform the calling >> equipment that the call has been forwarded to voicemail?" As far as I >> know, the answer to that is "no". >> > It's possible to detect call forwarding on ISDN. The telco must enable > it for you (for a large recurring fee when I last investigated). > Basically, you need to compare the actual connected number vs the number > that was dialed. I did a POC using an ISDN BRI line provided by Singtel. > My Asterisk setup at that time was Asterisk 1.0.3 with chan-capi. > > Hope this helps. > > > > > > ------------------------------ > > Message: 12 > Date: Mon, 22 May 2006 18:29:37 -0500 > From: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Office to Office via IAX2 problems > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii"; format=flowed > > Well, I changed host=ip address, qualify=2000, and rebooted computer > and now it connects. Hopefully, this will allow Office B to stay > connected to Office A. > > Thanks everyone. > > Doug > > At 03:30 PM 5/22/2006, you wrote: >>Hi Doug - >> >>>Office A routinely looses connection to Office B. When typing IAX2 >>>Show Peers, it will show as Unreachable. I issue IAX2 Reload and it >>>will work again for 1-3 days (haven't narrowed the time down yet). My >>>theory is that the DSL at Office2 is changing IP addresses regularly >>>and this is the cause of the problem??? This has been going on since >>>I set up Office B (2-3 weeks). I never had to touch Office B box. >>>Office B seemed to maintain connection, until now (see Issue 2). >> >>Just to cover all the bases. Can one machine talk to the other at >>all? Can you ssh from one box to another (if you don't use ssh, can >>you telnet to an open tcp port)? If not, it is surely a routing >>issue. >> >>If you can connect via non-asterisk methods, you might try increasing >>your qualify value to something higher (qualify=1500), or just remove >>it altogether for testing. It might be that the latency is high >>enough that the connection consistently fails to qualify. (What are >>the ping times, BTW?) >> >>I'll second Eric's advice to not use a DNS name for the host, even in >>your final setup. >> >>- Noah >> >> >>On 5/22/06, Lacy Moore - Aspendora <[EMAIL PROTECTED]> wrote: >>> >>>SInce you say it was working, I am assuming that both >>> officea.kicks-ass.net >>>and officeb.kicks-ass.net resolves to the real IP address and not an >>>internal address, correct? >>> >>>Also, are you providing DNS or someone else? Is this domain registered >>> to >>>you? I ask that because if it is not, and you are not providing DNS, it >>> may >>>be resolving to another IP address. But, since you said it is the same >>>using an IP address, this should not be the real issue. >>> >>>I'm not sure this would really have anything to do with it, but, if it >>> was >>>me, I would not have the two offices on the same subnet. I'd use >>> 192.168.1 >>>for one and 192.168.2 for the other. It just keeps things a little >>> simpler >>>routing wise. > > > > > ------------------------------ > > Message: 13 > Date: Tue, 23 May 2006 02:25:35 +0200 > From: <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Timeframe for QueueStatus values > To: <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > Hello all, > > I've a question regarding the values "completed" and "abandoned" that are > returned by the manager command "queuestatus". What is the timeframe for > these values, are they counted since the last asterisk boot, or per day, > or > is the timeframe configurable? > > Thanks and Regards > > Markus > > > > > > > > ------------------------------ > > Message: 14 > Date: Mon, 22 May 2006 20:57:44 -0400 > From: "BJ Weschke" <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Timeframe for QueueStatus values > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > On 5/22/06, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote: >> Hello all, >> >> I've a question regarding the values "completed" and "abandoned" that >> are >> returned by the manager command "queuestatus". What is the timeframe for >> these values, are they counted since the last asterisk boot, or per day, >> or >> is the timeframe configurable? >> > > Since the module was last loaded/reloaded. Therefore, there really > isn't a "static" timeframe. > > -- > Bird's The Word Technologies, Inc. > http://www.btwtech.com/ > > > ------------------------------ > > Message: 15 > Date: Mon, 22 May 2006 23:23:43 -0500 > From: Greg Oliver <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] CallerID > To: [email protected] > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain > > I am trying to set CIDNum to nothing, but my outgoing PRI controlled by > another PBX seems to fill in something when asterisk does not.. If I > set a number either in the sip channel for the phone, or from > extensions.con, it is realized.. If I try to leave them blank, or even > Not Defined, the main number of the pri gets sent out.. > > I am trying to debug a glitvh in or software and I need to be able to > make a test call with unknown (blank callerid).. I can successfully set > it to private, but that is not the same.. > > Any ideas? > > TIA > > -Greg > > > > ------------------------------ > > Message: 16 > Date: Mon, 22 May 2006 22:06:10 -0700 > From: Tom Lynn <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Help Avaya 4606 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=us-ascii > > The 4606 is a h.323 based phone. There is no SIP image to use with this > phone. > > On Fri, 12 May 2006 11:11:48 -0500, you wrote: > >>Hello all, >> >>I have asterisk working well with, Sipura, but I do not manage to form >>several phones avaya 4606, someone could have formed one avaya with >>asterisk? >> >>is it possible? >> >>update the firmware of the phone, but I do not achieve that it registers, >> >> >>I hope that someone could help me >> >>greetings to all >> >>Carlos Rojas > > > ------------------------------ > > Message: 17 > Date: Tue, 23 May 2006 15:18:07 +1000 > From: "Eric Bishop" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] US telco lingo > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > Could someone explain to a non-US dummy the following phrases I have seen > on > the list. > > "I can provide you with tier 1 termination 6/6. I can blend or NPANXX > breakout." > > "We provide US48 termination, blended rate for 1 MOU and above is .008 > with > 6/6." > > > What is 6/6? > > What is US48? > > What is blended? > > What is MOU? > > What is NPANXX breakout? > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20060522/d2718378/attachment-0001.htm > > ------------------------------ > > Message: 18 > Date: Mon, 22 May 2006 23:15:44 -0700 > From: "Kerry Garrison" <[EMAIL PROTECTED]> > Subject: RE: [Asterisk-Users] US telco lingo > To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > > > Could someone explain to a non-US dummy the following phrases I have > seen on the list. > > "I can provide you with tier 1 termination 6/6. I can blend or > NPANXX breakout." > > "We provide US48 termination, blended rate for 1 MOU and above is > .008 with 6/6." > > > What is 6/6? > 2/3 devil? > Normally I would take that to be minimum 6 second billiing and > billed in 6 seconds increments. > > What is US48? > Contentinal US, lower 48, all states by Alaska and Hawaii. > > What is blended? > What you do with ice, alchohol, and a mixer > > > > > > ------------------------------ > > Message: 19 > Date: Tue, 23 May 2006 17:23:08 +1000 > From: Warrick Zedi <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Faxing - machines stop talking, line stays > up > To: [email protected] > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain > > Hi, > > I haven't seen this situation described anywhere. We have an Asterisk > server configured with 2 T4XXP cards and a TDM400P. We are using > spandsp's txfax to send faxes on one of the E1 channels. > > The calls originates and the faxes start talking to each other for about > 10 seconds. Then they just stop. The line stays up for a while after > that until the remote end hangs up. The local line stays up for a little > while longer and then eventually hangs up. > > I've fiddled with the gain settings in zapata.conf using ztmonitor as a > guide to no avail. I've also tried busydetect=no in zapata.conf. > > In addition, changing the txgain appears to have no effect (going by > ztmonitor), while rxgain does have an effect. > > Any suggestions welcome. > > Cheers, > Warrick > > > > ------------------------------ > > Message: 20 > Date: Tue, 23 May 2006 09:27:28 +0200 > From: "Alejandro Vargas" <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] FAX and Asterisk > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > 2006/5/23, Rene Nelson <[EMAIL PROTECTED]>: >> >> I want to accept faxes via SIP/IAX2 (yes I've read the posts that it >> isnt >> reccomended). My PBX is 100% Virtual with the exception of one IAX > > I made it with iaxmodem and hylafax. It can route fax to email > converting fax to pdf, and many things you like to do, based on the > caller id i.e. > > > -- > Alejandro Vargas > > > ------------------------------ > > Message: 21 > Date: Tue, 23 May 2006 08:35:55 +0100 (BST) > From: John Joseph <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] TDM400P , "ztcfg ?vv error ", "Does it have > to do with my PC hardware ?" > To: Asterisk Users <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=iso-8859-1 > > Hi > I have a working asterisk with TDM400P card , > today I was trying to install asterisk with another > TDM400P card in another machine, I copied the > zapata.conf and zaptel.conf of the working asterisk > To my surprise I am getting this error > when I run ztcfg -vv > > Changing signalling on channel 1 from Clear channel to > FXS Kewlstart > ZT_CHANCONFIG failed on channel 1: Invalid argument > (22) > Did you forget that FXS interfaces are configured with > FXO signalling > and that FXO interfaces use FXS signalling? > > I would like to know , whether this error is coming > , because of my second PC hardware , I want to > confirm really it is my PC problem , or not with my > configuration files > Guidance requested > Thanks > Joseph John > > > Send instant messages to your online friends http://uk.messenger.yahoo.com > > > ------------------------------ > > Message: 22 > Date: Tue, 23 May 2006 17:40:09 +1000 > From: Warrick Zedi <[EMAIL PROTECTED]> > Subject: [Fwd: [Asterisk-Users] Faxing - machines stop talking, line > stays up] > To: [email protected] > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > Sorry, > > I didn't mention we're using spandsp txfax (0.0.2pre25) to send the > faxes. > -------------- next part -------------- > An embedded message was scrubbed... > From: Warrick Zedi <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Faxing - machines stop talking, line stays up > Date: Tue, 23 May 2006 17:23:08 +1000 > Size: 4568 > Url: > http://lists.digium.com/pipermail/asterisk-users/attachments/20060523/68c2113d/attachment-0001.eml > > ------------------------------ > > Message: 23 > Date: Tue, 23 May 2006 09:58:11 +0200 > From: "Giordano Grandis" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] SIP session number > To: <[email protected]> > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="iso-8859-1" > > Hi all, > just a question: how can i known the number of SIP session? > In general and not for a single user. > > Thanks > > Giordano > > Le informazioni contenute nella presente e-mail e nei documenti > eventualmente allegati possono essere confidenziali e sono comunque > riservate al destinatario della stessa. La loro diffusione, distribuzione > e/o copiatura da parte di terzi รจ proibita. Se avete ricevuto questa > comunicazione per errore, Vi preghiamo di informare immediatamente il > mittente del messaggio e di distruggere questa e-mail. > > This e-mail may contain confidential and/or privileged information. If you > are not the intended recipient (or have received this e-mail in error) > please notify the sender immediately and destroy this e-mail. Any > unauthorised copying, disclosure or distribution of the material in this > e-mail is strictly forbidden. > > > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20060523/ebc152b4/attachment-0001.htm > > ------------------------------ > > Message: 24 > Date: Tue, 23 May 2006 10:10:31 +0200 > From: "Asterisk" <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Logger rotate & master.csv > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Message-ID: > <[EMAIL PROTECTED]> > Content-Type: text/plain; charset="us-ascii" > > Hi guys, > > > > I have noticed that 'logger rotate' command only rotates log files in > the /var/log/asterisk directory, but not in the subdirectories. How > could I rotate my /var/log/asterisk/cdr-custom/Master.csv log file? > > > > Regards, > > Alex > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-users/attachments/20060523/c4518a1c/attachment-0001.htm > > ------------------------------ > > Message: 25 > Date: Tue, 23 May 2006 10:11:55 +0200 > From: Kai Ober <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Free/Open pci telco card > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=ISO-8859-15; format=flowed > > Hi List, > > While I was surfing the net last week, > I found a link for "open source" pci telco cards. > I'm not sure if it were isdn or analog related. > > The layout an all the stuff was free downloadable, so that you can build > your own cards. > > Does anybody have the link? > > Yes, I know there is google, but i searched for over an hour, but can't > find anything. > maybe i use the wrong search words, anny suggestions? > > thx > Kay > > > > ------------------------------ > > Message: 26 > Date: Tue, 23 May 2006 10:14:14 +0200 > From: Louis-David Mitterrand <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] Re: voicemail access on the Thomson ST2030 ? > To: picciuX <[EMAIL PROTECTED]> > Cc: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=us-ascii > > On Mon, May 22, 2006 at 12:25:34PM +0200, picciuX wrote: >> for provisioning files to be taken, you have to change the "config_sn" >> parameter each time you modify the file, otherwise the phone assumes >> nothing >> has changed. > > Even after a factory reset of the phone? (ie: power-cycle with > speaker+mute buttons pressed) > > Thanks, > > > ------------------------------ > > Message: 27 > Date: Tue, 23 May 2006 10:21:05 +0200 (CEST) > From: Remco Barende <[EMAIL PROTECTED]> > Subject: [Asterisk-Users] A call from a call file always does a > redial? > To: Asterisk Users List <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: TEXT/PLAIN; charset=US-ASCII; format=flowed > > > I have an issue with the Snom 360's (any firmware) and asterisk call > files. When you setup a call using a call file from Asterisk and the call > is connected, Asterisk will start to redial the call after about 5 minutes > when the conversation is already ongoing. (Annoying and it can only be > avoided by disabling call waiting) > > I tried to reproduce the problem with a GrandStream phone and a Sipura > ATA, it doesn't occur. > > I guess these are 2 problems : > > 1) The callfile specifies that a call should not be retried, still * does > a redial > > 2) I *guess* the Snom is returning a different signal than other phones > when the call is answered up making Asterisk believe that the call > never succeeded. > > I registered this as a bug in mantis previously but nobody was able to > reproduce, I know found out that it is only happening when using a > Snom 360 as client. > > > > ------------------------------ > > Message: 28 > Date: Tue, 23 May 2006 10:23:15 +0200 > From: Philipp Ott <[EMAIL PROTECTED]> > Subject: Re: [Asterisk-Users] Deadlocks in 1.2.7.1 > To: Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]> > Message-ID: <[EMAIL PROTECTED]> > Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed > > Hello! > > Am 17.05.2006 um 13:05 schrieb Philipp Ott: > >> Hello! >> >> Unfortunately we are seeing lately (2-3 times during a day) that >> asterisk seems to hang up somehow - no new calls can be made and >> sip show peers and other commands show no obvious problem. We then >> recompiled 1.2.7.1 with all the DEBUG_ turned on in the makefile >> and now we see the following messages: >> >> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236 >> __ast_pthread_mutex_lock: chan_sip.c line 3116 (sip_alloc): >> Deadlock? waited 460 sec for mutex '&iflock'? >> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:239 >> __ast_pthread_mutex_lock: chan_sip.c line 11257 (do_monitor): >> '&iflock' was locked here. >> May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:236 >> __ast_pthread_mutex_lock: pbx.c line 2017 >> (ast_extension_state_del): Deadlock? waited 460 sec for mutex >> '&hintlock'? >> May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:239 >> __ast_pthread_mutex_lock: pbx.c line 1892 (ast_hint_state_changed): >> '&hintlock' was locked here. >> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236 >> __ast_pthread_mutex_lock: chan_sip.c line 3116 (sip_alloc): >> Deadlock? waited 460 sec for mutex '&iflock'? >> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:239 >> __ast_pthread_mutex_lock: chan_sip.c line 11257 (do_monitor): >> '&iflock' was locked here. >> May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:236 >> __ast_pthread_mutex_lock: pbx.c line 2017 >> (ast_extension_state_del): Deadlock? waited 460 sec for mutex >> '&hintlock'? >> May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:239 >> __ast_pthread_mutex_lock: pbx.c line 1892 (ast_hint_state_changed): >> '&hintlock' was locked here. >> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236 >> __ast_pthread_mutex_lock: chan_sip.c line 3116 (sip_alloc): >> Deadlock? waited 460 sec for mutex '&iflock'? >> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:239 >> __ast_pthread_mutex_lock: chan_sip.c line 11257 (do_monitor): >> '&iflock' was locked here. >> May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:236 >> __ast_pthread_mutex_lock: pbx.c line 2017 >> (ast_extension_state_del): Deadlock? waited 460 sec for mutex >> '&hintlock'? >> May 17 06:46:05 ERROR[8625]: include/asterisk/lock.h:239 >> __ast_pthread_mutex_lock: pbx.c line 1892 (ast_hint_state_changed): >> '&hintlock' was locked here. >> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:236 >> __ast_pthread_mutex_lock: chan_sip.c line 3116 (sip_alloc): >> Deadlock? waited 460 sec for mutex '&iflock'? >> May 17 06:46:05 ERROR[8606]: ../include/asterisk/lock.h:239 >> __ast_pthread_mutex_lock: chan_sip.c line 11257 (do_monitor): >> '&iflock' was locked here. >> >> This continues until someone stops asterisks and restarts it. > > > Stepping back to version 1.2.4 solves the problem of a hanging > asterisk, however occassionally we see 5-15 seconds runs of > > May 23 00:28:35 DEBUG[3212] chan_sip.c: Failed to grab lock, trying > again... > > messages in the log file and during this time no call processing > happens. Then asterisk recovers from this locking state and > continues. 1.2.7.1 hangs in there forever. > > Any clues as to why this happens? > > Regards > Philipp Ott > > > > ------------------------------ > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > End of Asterisk-Users Digest, Vol 22, Issue 125 > *********************************************** > > _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
