Re: [Asterisk-Users] Re: Record inbound and outbound calls to and from one number.
I wouldn't have asked if it didn't matter. All I want is a place where I can find the regulations. I already know how to use the Playback command. On Mon, 2005-01-31 at 06:45 +, Tom Shoval wrote: Tim Mattison wrote: Good call. For our American readers... does anyone know where I can obtain a list of states/counties and their regulations in regards to call recording? On Sun, 2005-01-30 at 22:10 +, Mike Dent wrote: or maybe country? or should that be County? :) Mike does it matter? you should provide warning to everybody. you can do it in your top menu, by stating that some calls are monitored for QA purposes, like most call centers do these days anyway. ___ 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
Re: [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 crashes with Ouch ... error while writing audio data: : Broken pipe
If the problem is in mpg123 than way just not to replace it? Here is one very good example how to do it: http://www.voip-info.org/wiki-Asterisk+mpg123+faking+it but I think that the problem maybe is coming from the BRIstuffed * - the patches and etc. Lubo - AppRadius Project: Full RADIUS AAA support for Asterisk PBX http://appradius.minitelecom.org/ - Remco Barende wrote: On Sun, 30 Jan 2005, Martin List-Petersen wrote: Citat Remco Barende [EMAIL PROTECTED]: I have Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 up and running. Everything seems to be running fine but after some time asterisk just goes crazy (even withouth any incoming or outgoing call activity perviously). If I leave the box up for some time * goes haywire and the console is flooded with this message: Ouch ... error while writing audio data: : Broken pipe At that time I can see that there are multiple instances of mpg123 active. The solution to this problem is to kill-9 mpg123, do the same for *, unload the modules and then load the modules again and start asterisk. If I do not unload re-load the modules I cannot access the ISDN line nor do incoming calls work. I really don't know where to look for this problem. Is it possible to completely disable music on hold? Asterisk combined mpg123 is causing nothing but problems anyway, the current stable still leaves abandoned mpg123 processes. It doesn't work :( Asterisk doesn't go haywire flooding the console but now simply bombs out with : *CLI Segmentation fault I guess that qualifies it as a bristuff bug? ___ 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
RE: [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 crasheswith Ouch ... error while writing audio data: : Broken pipe
Have the same problem with PRI, Probably the problem is in asterisk MusicOnHold feature. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Remco Barende Sent: Monday, January 31, 2005 8:33 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 crasheswith Ouch ... error while writing audio data: : Broken pipe On Sun, 30 Jan 2005, Martin List-Petersen wrote: Citat Remco Barende [EMAIL PROTECTED]: I have Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 up and running. Everything seems to be running fine but after some time asterisk just goes crazy (even withouth any incoming or outgoing call activity perviously). If I leave the box up for some time * goes haywire and the console is flooded with this message: Ouch ... error while writing audio data: : Broken pipe At that time I can see that there are multiple instances of mpg123 active. The solution to this problem is to kill-9 mpg123, do the same for *, unload the modules and then load the modules again and start asterisk. If I do not unload re-load the modules I cannot access the ISDN line nor do incoming calls work. I really don't know where to look for this problem. Is it possible to completely disable music on hold? Asterisk combined mpg123 is causing nothing but problems anyway, the current stable still leaves abandoned mpg123 processes. It doesn't work :( Asterisk doesn't go haywire flooding the console but now simply bombs out with : *CLI Segmentation fault I guess that qualifies it as a bristuff bug? ___ 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
Re: [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 crashes with Ouch ... error while writing audio data: : Broken pipe
Hi, please start asterisk -vvvcg (so it creates a core file when it segfaults), then run gdb /usr/sbin/asterisk corefile, hit Enter a few times and run a backtrace using bt. Please email the output. I doubt that it's bristuff bug, since many users have already successfully upgraded. best regards Klaus Am Montag, den 31.01.2005, 08:33 +0100 schrieb Remco Barende: On Sun, 30 Jan 2005, Martin List-Petersen wrote: Citat Remco Barende [EMAIL PROTECTED]: I have Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 up and running. Everything seems to be running fine but after some time asterisk just goes crazy (even withouth any incoming or outgoing call activity perviously). If I leave the box up for some time * goes haywire and the console is flooded with this message: Ouch ... error while writing audio data: : Broken pipe At that time I can see that there are multiple instances of mpg123 active. The solution to this problem is to kill-9 mpg123, do the same for *, unload the modules and then load the modules again and start asterisk. If I do not unload re-load the modules I cannot access the ISDN line nor do incoming calls work. I really don't know where to look for this problem. Is it possible to completely disable music on hold? Asterisk combined mpg123 is causing nothing but problems anyway, the current stable still leaves abandoned mpg123 processes. It doesn't work :( Asterisk doesn't go haywire flooding the console but now simply bombs out with : *CLI Segmentation fault I guess that qualifies it as a bristuff bug? ___ 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
Re: [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 crashes with Ouch ... error while writing audio data: : Broken pipe
Yes, I do think the segfault problem is in bristuff. I just noticed another problem though! If there is one ongoing conversation and a new call is coming in, as soon as the new call is answered, the call that was going on goes dead on one end (the other party can hear the person callingfrom the * end but the reverse audio is not transmitted). I think these are bugs in bristuff RC5. On Mon, 31 Jan 2005, Lubomir Christov wrote: If the problem is in mpg123 than way just not to replace it? Here is one very good example how to do it: http://www.voip-info.org/wiki-Asterisk+mpg123+faking+it but I think that the problem maybe is coming from the BRIstuffed * - the patches and etc. Lubo - AppRadius Project: Full RADIUS AAA support for Asterisk PBX http://appradius.minitelecom.org/ - Remco Barende wrote: On Sun, 30 Jan 2005, Martin List-Petersen wrote: Citat Remco Barende [EMAIL PROTECTED]: I have Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 up and running. Everything seems to be running fine but after some time asterisk just goes crazy (even withouth any incoming or outgoing call activity perviously). If I leave the box up for some time * goes haywire and the console is flooded with this message: Ouch ... error while writing audio data: : Broken pipe At that time I can see that there are multiple instances of mpg123 active. The solution to this problem is to kill-9 mpg123, do the same for *, unload the modules and then load the modules again and start asterisk. If I do not unload re-load the modules I cannot access the ISDN line nor do incoming calls work. I really don't know where to look for this problem. Is it possible to completely disable music on hold? Asterisk combined mpg123 is causing nothing but problems anyway, the current stable still leaves abandoned mpg123 processes. It doesn't work :( Asterisk doesn't go haywire flooding the console but now simply bombs out with : *CLI Segmentation fault I guess that qualifies it as a bristuff bug? ___ 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
Re: [Asterisk-Users] Zap channels in AU hanging up on STD pips
On Mon, 2005-01-31 at 16:51, jurgen wrote: Hi Howard, Which provider are you with? We're with Primus Business here in Melbourne, and haven't had anything like what you're describing. For reference, here's a snip of my zapata.conf: Big T [channels] language=en context=local signalling=fxs_ks usecallerid=no echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=5 Sometimes the busydetect hack hits a false positive and disconnects during a conversation, so I'm thinking of upping the busycount, but aside from that, calls through this are quite reliable. Mine's pretty similar: context = default signalling = fxs_ks echocancel = 128 echocancelwhenbridged = yes echotraining = yes relaxdtmf = yes ;pulsedial = yes pulsedial = no rxgain = +15% txgain = +5% immediate = no busydetect = yes busycount = 5 callprogress = yes musiconhold = default usecallerid = yes callerid = asreceived ;usedistinctiveringdetection = yes useincomingcalleridonzaptransfer = yes faxdetect = both group = 1 channel = 4 Best, ...jurgen On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Japan
Has anyone tried Sipura products such as the 3000 in Japan? Jason Steven Critchfield wrote: On Mon, 2005-01-31 at 12:53 +0900, Jason Frisch wrote: Sorry for my ignorance, but what is J1? I actually hope to use Softbanks fiber-based IPtel service, but I believe they require VoIP TA so I guess the end result is just a standard analog line. J1 is a Japanese T1 or close equivalent. If IPTel uses a VoIP TA(voice over IP terminal adapter), you might want to check into the type of signaling they use. It sounds like they might be using SIP, H323, or MGCP to deliver the service. In that case you might be able to swap the TA for asterisk with no or minimal trouble. Then you are free to provision your side of the link as you wish. ___ 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] Call Screen Macro Not Exiting when call rejected
Thanks everyone for your help. The code in the dialplan was ok. I had to switch to CVS head and everything worked straight away. Any clues on when this will be working in the stable release ? Thanks ___ 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] Zap channels in AU hanging up on STD pips
Try busydetect=no Simon Brown -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes Sent: Monday, 31 January 2005 19:17 To: jurgen; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Zap channels in AU hanging up on STD pips On Mon, 2005-01-31 at 16:51, jurgen wrote: Hi Howard, Which provider are you with? We're with Primus Business here in Melbourne, and haven't had anything like what you're describing. For reference, here's a snip of my zapata.conf: Big T [channels] language=en context=local signalling=fxs_ks usecallerid=no echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=5 Sometimes the busydetect hack hits a false positive and disconnects during a conversation, so I'm thinking of upping the busycount, but aside from that, calls through this are quite reliable. Mine's pretty similar: context = default signalling = fxs_ks echocancel = 128 echocancelwhenbridged = yes echotraining = yes relaxdtmf = yes ;pulsedial = yes pulsedial = no rxgain = +15% txgain = +5% immediate = no busydetect = yes busycount = 5 callprogress = yes musiconhold = default usecallerid = yes callerid = asreceived ;usedistinctiveringdetection = yes useincomingcalleridonzaptransfer = yes faxdetect = both group = 1 channel = 4 Best, ...jurgen On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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
Re: [Asterisk-Users] Monitor calls timeout
jurgen wrote: Thanks for the suggestion, but it's no good. It still times out after 10 seconds. It seems to be something in the Monitor application, rather than anywhere else. I can playback a sound (like the monkeys, or MOH) forever and ever without timing out. Monitoring kills itself though. That's because * is getting tired of waiting for the caller to dial an extension. Try this exten = s,1,Answer exten = s,2,Monitor(wav,testrecord,m) exten = s,3,Wait(600) exten = s,4,Goto(s,3) ___ 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] Zap channels in AU hanging up on STD pips
On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. I had the same problem (before I switched to Telstra ISDN). Increasing busycount to 8 fixed it. -Shaun Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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] Instant Messaging
Can Asterisk work as Instant Messaging Proxy? Is there anybody who can help me?! Best regards from Italy ___ 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] IAX2 firmware for PA168x (Giptel G100, Siptronic ST-100 etc)
On Sun, 30 Jan 2005 21:40:01 +0100, Philipp von Klitzing wrote: Hi there, this is just a short note about one of the PA168x based phones out there which I obtained as Giptel G100 (aka Siptronic ST-100): For some reason this phone would refuse to register with Asterisk using SIP, but after uploading the IAX2 firmware instead it finally came to life: http://www.voip-info.org/tiki-index.php?page=GIPTEL+IP+Phones Cheers, Philipp except it should be 5060. I current use the WuChuan version (5111soft.com) SIP works well, (so does IAX2). correct port addressing and rtp ports is required. . ___ 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] NAT and SIP
Hi, Does Asterisk have a limit to how many NAT'ed SIP clients it supports behind a single IP? I have the weirdest problem ever. I have three SIP endpoints. SNOM phones, if it matters. Their extensions are 200, 201 and 202. Apart from the username/password, the sip entries in sip.conf all have identical configuration. They're all NAT'ed behind the same IP. 200 and 202 registers just fine, but 201 is completely ignored by Asterisk. I've traced the REGISTER packets from the phones and compared 202 to 201. They're pretty much identical, apart from tags, CSEQ and stuff like that. 202 gets a 100 Trying reply, but 201 doesn't get anything. There's nothing going on in Asterisk console debug output. I then moved the 201 phone to a different LAN, so it got NAT'ed behind a different IP. There are other phones on that LAN which registers fine. Still no response from Asterisk though. Then I moved it to a third network, still NAT'ed, but without any other SIP clients. There it registered just fine. I then disconnected it, let it time out in Asterisk and connected it to the first LAN again. No reply. So this leads me to believe there's some kind of limit per IP on NAT'ed SIP clients. Can anybody shed some light on this? [200] type= friend username= 200 secret = 200secrets host= dynamic amaflags= default accountcode = myrealm context = incoming realm = myrealm dtmfmode= rfc2833 language= da nat = yes callgroup = 20 pickupgroup = 20 callerid= SNOM 200 qualify = 3000 [201] type= friend username= 201 secret = 201secrets host= dynamic amaflags= default accountcode = myrealm context = incoming realm = myrealm dtmfmode= rfc2833 language= da nat = yes callgroup = 20 pickupgroup = 20 callerid= SNOM 201 qualify = 3000 [202] type= friend username= 202 secret = 202secrets host= dynamic amaflags= default accountcode = myrealm context = incoming realm = myrealm dtmfmode= rfc2833 language= da nat = yes callgroup = 20 pickupgroup = 20 callerid= SNOM 202 qualify = 3000 -- Regards, Tais M. Hansen ComX Networks A/S Tel: +45-70257474 Fax: +45-70257374 pgp6srYq0CAf3.pgp Description: 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
Re: [Asterisk-Users] Zap channels in AU hanging up on STD pips
Guys, On a telsttra line you can have the STD pip's removed, I recently did this on a few on my lines. If you want the setting to ask telstra for, ask me off list and i'll try and find it. On Mon, 31 Jan 2005 16:51:38 +1100, jurgen wrote: Hi Howard, Which provider are you with? We're with Primus Business here in Melbourne, and haven't had anything like what you're describing. For reference, here's a snip of my zapata.conf: [channels] language=en context=local signalling=fxs_ks usecallerid=no echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=5 Sometimes the busydetect hack hits a false positive and disconnects during a conversation, so I'm thinking of upping the busycount, but aside from that, calls through this are quite reliable. Best, ...jurgen On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- [EMAIL PROTECTED] is jurgen's gmail address. Visit http://jurgen.ca/ for more yummy goodness. ___ 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
RE: [Asterisk-Users] Call Waiting Audio Prompt
Thanks for the replies everyone how do you expect to get the indication that you have a callwaiting call? The whole point is I don't want it. The beep is a guard that hides the caller-id fsk spill also. So you can't get callwaiting-callerid and not have a beep. I don't really need that either, users can see whos waiting on hold for them via the web page so anything caller waiting related is fine but only as long as it doesn't have negative impact on the call quality. Setting the indication durations to zero has helped hugely as the sound is far less intrusive now. Also all of the end points are essentially SIP phones (or DECT phones plugged into 2102's) so callerID doesn't need to be passed inbound. I will try what Jon sugggests: Zaptel.conf callwaiting=no callwaitingcallerid=no It didn't really occur to me as was looking at configuring the SIP side. This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. ___ 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] Zap channels in AU hanging up on STD pips
witht he bigt just ask provisioningto have NOPIPS set on the required lines. simple really On Mon, 31 Jan 2005 19:17:05 +1100, Howard Lowndes wrote: On Mon, 2005-01-31 at 16:51, jurgen wrote: Hi Howard, Which provider are you with? We're with Primus Business here in Melbourne, and haven't had anything like what you're describing. For reference, here's a snip of my zapata.conf: Big T [channels] language=en context=local signalling=fxs_ks usecallerid=no echocancel=yes echocancelwhenbridged=yes busydetect=yes busycount=5 Sometimes the busydetect hack hits a false positive and disconnects during a conversation, so I'm thinking of upping the busycount, but aside from that, calls through this are quite reliable. Mine's pretty similar: context = default signalling = fxs_ks echocancel = 128 echocancelwhenbridged = yes echotraining = yes relaxdtmf = yes ;pulsedial = yes pulsedial = no rxgain = +15% txgain = +5% immediate = no busydetect = yes busycount = 5 callprogress = yes musiconhold = default usecallerid = yes callerid = asreceived ;usedistinctiveringdetection = yes useincomingcalleridonzaptransfer = yes faxdetect = both group = 1 channel = 4 Best, ...jurgen On Mon, 31 Jan 2005 16:34:38 +1100, Howard Lowndes [EMAIL PROTECTED] wrote: Is anyone having/had a problem with a TDM400P card hanging up on STD outbound calls as soon as the called party answers. I'm guessing that * is responding to the STD pips in some way. -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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
Re: [Asterisk-Users] TE410P card in an HP-Compaq DL380 G4 server
Did anyone get anywhere with this thread? Any HP G4 series servers working? On Wed, 26 Jan 2005 09:46:31 +1100, Eric Bishop [EMAIL PROTECTED] wrote: Has anyone had any luck with this issue and new Asterisk/Zaptel releases (1.05/1.04)? I am still searching for a solution and waiting for that Eureka! moment.. On Thu, 20 Jan 2005 09:20:09 +0100, Tais M. Hansen [EMAIL PROTECTED] wrote: On Wednesday 19 January 2005 23:15, Eric Bishop wrote: Well guys this is truly bizarre. I managed to get a DL360 G3 to show interrupts with FC2 but not FC3. Exact same config and setup proceedure. Ofcourse neither FC2 or FC3 show interrupts with the DL360 G4. I think TE410P is just a flakey card. Anyone got a DL360 G3 going with a TE410P and FC3? I did manage to get a TE110P running on the DL380 G4. Still can't get the TE410P working in the G4 though. Supports your theory. Sadly we're now being forced to look elsewhere for PRI cards. -- Regards, Tais M. Hansen ComX Networks A/S Tel: +45-70257474 Fax: +45-70257374 ___ 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] PRI for Data and Voice
Do you have a config sample on how to handle digital PPP sessions in Asterisk? On Sat, 29 Jan 2005 15:16:51 +0100 (CET), Peter Svensson [EMAIL PROTECTED] wrote: On Sat, 29 Jan 2005, David Norton wrote: Currently I only have 1 PRI which I am using for dial-in customers. The line is connected to a Portmaster3. I have never used more than 10 concurrent channels. The calls can be both analog or ISDN. It would be a waste to order another PRI for my Asterisk box. Is there any way of splitting a PRI into 2 PRI's of 15 channels each, or plugging the PRI into the * box and it send the data calls to the portmaster, or handles them itself? Off the top of my head I can think of a few solutions: * Use a multiport T1 zapata card (TE405P or TE410P) and connect your systems this way: PSTN -PRI- Asterisk -PRI- Portmaster \ ---lan--- voip stuff With suitable parameters to the Dial application in Asterisk the forwarded calls will be passed transparently between the interfaces. This is similar to how we handle isdn data calls. * The zapata driver can handle digital but not analog ppp connections in the driver. If you wanted to you could use the above solution but have the Asterisk box handle the digital data calls. Not much is gained since you still need the Portmaster for the analog data calls. * Use a pri card with an on board DSP in the Asterisk box. The so called active isdn cards are usually so equipped. Cards in this category are the Eicon Diva Server T1/PRI cards (or the E1 equivalent) and probably more. I think they all interface to Asterisk via CAPI. * Buy a box with a dedicated box that does both VoIP and data call termination and interface to Asterisk via VoIP. Peter ___ 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] congestion problem with only one number
Hi all,
I have this weird problem.
I'm running asterisk 1.0.3 on Debian Sid (official debian package).
We have 2 fritz ISDN cards.
All is working great.
Till I called the bank. It rings one time and then gives me
the congestion tone.
Here is what I see on the CLI (phone nr obfuscated for
privacy reasons):
-- Executing Dial(SCCP/michiel-0004, Modem/g1:xx|50|Ttr) in
new stack
-- Called g1:0342426530
-- Asked to indicate '3' (Dialing) condition on channel
SCCP/michiel-0004
-- Current tone (36) is equiv to wanted tone (36). Ignoring.
-- Modem[i4l]/ttyI3 is busy
-- Hungup 'Modem[i4l]/ttyI3'
== Everyone is busy/congested at this time
-- Sending tone 127
-- Executing Congestion(SCCP/michiel-0004, ) in new stack
I only have this with the bank.
Is it possible there is some PBX at the bank that messes up
normal call progress in * ???
This is a Dutch bank, maybe ppl in Holland using * can try
to call the bank. It is the rabobank.
relevant configs:
modem.conf
[interfaces]
context=remote
driver=i4l ; isdn4linux - an alternative to i4l is to use chan_capi
language=en
type=autodetect
stripmsd=0
dialtype=tone
mode=answer
group=1 ; group=1,2,3,9-12
msn=x
incomingmsn=*
device = /dev/ttyI0
device = /dev/ttyI1
device = /dev/ttyI2
device = /dev/ttyI3
The lines in extensions.conf that handle outgoing calls
exten = _0X,1,Dial(${TRUNK}:${EXTEN},50,Ttr)
exten = _0X,2,congestion
exten = _00X.,1,Dial(${TRUNK}:${EXTEN},50,Ttr)
exten = _00X.,2,congestion
exten = _4X,1,Dial(${TRUNK}:0342${EXTEN},50,Ttr)
exten = 112,1,Dial(${TRUNK}:112,50,Ttr)
exten = _0[89]00X.,1,Dial(${TRUNK}:${EXTEN},50,Ttr)
exten = _0[89]00X,2,congestion
Greetz,
Michiel
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Re: [Asterisk-Users] Asterisk Prepaid Application Help
You just have to define the accountnumber into your sip.conf/iax.conf. Of course, accountnumber = card number of areskicc! Then the IVR application wont prompt anymore to enter the cardnumber and ask directly to dial the destination numer. Hope this is help, /Areski On Sat, 2005-01-29 at 05:23, Ritesh Jalan wrote: I have running Asterisk latest version with areskiCC as a prepaid application I like to have a prepaid application running for sip users, and in that instead of sip user dialing the account no. it should take the account no. from its user account, how to do that?? Thanks Regards Ritesh Jalan __ ___ 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] Group Extension
Hello, i got a question, i need to create a group extension, to make calls to 6 sw phones, but i need to know if asterisk can do help me to get a unique number and check what extension has received less calls than the others, and pass the new call. We got a call center and want to know if we can distribute the calls depending in what extension is available and from the extensions that are available pass the call to the operator that has answered less calls, can i do this with *? can i get statistics from the use for an extension? can anybody help me?? TIA Edgar ___ 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] Indication of transfer on display
Hi all, I'm using asterisk 1.0.2 (the Debian Sarge package) with Cisco 7905G phones (SIP firmware). I've defined a macro to do some custom CallerID stuff for our 100-number ISDN range (so we can see what line they've called). What I'd like to do is have the phone display update (to the original called number?) when an incoming call is transferred from one phone to another. At present there's no way to tell, short of looking at the asterisk console for when the channel hangs up - the phone still displays the internal caller ID of the first phone. I see that the wiki shows this features.conf option for CVS HEAD, which would almost do the job: xfersound = beep ; to indicate an attended transfer is complete However, am I correct in assuming that this only applies to transfers done with the # key, rather than the phone's own transfer function? Any suggestions? Thanks, Andrew -- Linux supports the notion of a command line or a shell for the same reason that only children read books with only pictures in them. Language, be it English or something else, is the only tool flexible enough to accomplish a sufficiently broad range of tasks. -- Bill Garrett ___ 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] NAT and SIP
On Monday 31 January 2005 09:29, Tais M. Hansen wrote: Hi, Does Asterisk have a limit to how many NAT'ed SIP clients it supports behind a single IP? [...] Theoretical limit is around 65536 clients. B ___ 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] Group Extension
Hi,Edgar, Config the agents.conf correctly and it will do what you want. For more information, search it in the wiki please. Regards. David http://www.iaxtalk.com - Original Message - From: Edgar de Leon [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Monday, January 31, 2005 5:41 PM Subject: [Asterisk-Users] Group Extension Hello, i got a question, i need to create a group extension, to make calls to 6 sw phones, but i need to know if asterisk can do help me to get a unique number and check what extension has received less calls than the others, and pass the new call. We got a call center and want to know if we can distribute the calls depending in what extension is available and from the extensions that are available pass the call to the operator that has answered less calls, can i do this with *? can i get statistics from the use for an extension? can anybody help me?? TIA Edgar ___ 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
Re: [Asterisk-Users] Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 crashes with Ouch ... error while writing audio data: : Broken pipe
This is the output of gdb:
Reading symbols from /usr/lib/asterisk/modules/cdr_pgsql.so...done.
Loaded symbols for /usr/lib/asterisk/modules/cdr_pgsql.so
Reading symbols from /usr/lib64/libpq.so.3...done.
Loaded symbols for /usr/lib64/libpq.so.3
Reading symbols from /lib64/libcrypt.so.1...done.
Loaded symbols for /lib64/libcrypt.so.1
Reading symbols from /lib64/libnsl.so.1...done.
Loaded symbols for /lib64/libnsl.so.1
Reading symbols from /usr/lib/asterisk/modules/chan_sccp.so...done.
Loaded symbols for /usr/lib/asterisk/modules/chan_sccp.so
Reading symbols from /lib64/libgcc_s.so.1...done.
Loaded symbols for /lib64/libgcc_s.so.1
#0 sccp_pbx_read (ast=0x0) at sccp_pbx.c:38
38if (f-frametype == AST_FRAME_VOICE) {
(gdb)
(gdb)
(gdb)
(gdb)
(gdb) bt
#0 sccp_pbx_read (ast=0x0) at sccp_pbx.c:38
#1 0x00416261 in ast_read (chan=0x6439f0) at channel.c:1337
#2 0x0041aa42 in ast_waitfordigit (c=0x6439f0, ms=2) at
channel.c:1140
#3 0x002a9e326af1 in sccp_start_channel (data=0x0) at sccp_pbx.c:505
#4 0x002a95774c6b in start_thread () from /lib64/tls/libpthread.so.0
#5 0x002a95e8ce43 in thread_start () from /lib64/tls/libc.so.6
#6 0x in ?? ()
Asterisk is bombing out on chan_sccp?
Thanks!
On Mon, 31 Jan 2005, Klaus-Peter Junghanns wrote:
Hi,
please start asterisk -vvvcg (so it creates a core file when it
segfaults), then run gdb /usr/sbin/asterisk corefile, hit
Enter a few times and run a backtrace using bt. Please email
the output. I doubt that it's bristuff bug, since many users
have already successfully upgraded.
best regards
Klaus
Am Montag, den 31.01.2005, 08:33 +0100 schrieb Remco Barende:
On Sun, 30 Jan 2005, Martin List-Petersen wrote:
Citat Remco Barende [EMAIL PROTECTED]:
I have Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 up and running. Everything
seems to be running fine but after some time asterisk just goes crazy
(even withouth any incoming or outgoing call activity perviously).
If I leave the box up for some time * goes haywire and the console is
flooded with this message:
Ouch ... error while writing audio data: : Broken pipe
At that time I can see that there are multiple instances of mpg123 active.
The solution to this problem is to kill-9 mpg123, do the same for *,
unload the modules and then load the modules again and start asterisk. If
I do not unload re-load the modules I cannot access the ISDN line nor do
incoming calls work.
I really don't know where to look for this problem. Is it possible to
completely disable music on hold? Asterisk combined mpg123 is causing
nothing but problems anyway, the current stable still leaves abandoned
mpg123 processes.
It doesn't work :( Asterisk doesn't go haywire flooding the console but
now simply bombs out with :
*CLI
Segmentation fault
I guess that qualifies it as a bristuff bug?
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Re: [Asterisk-Users] Group Extension
Edgar de Leon wrote: Hello, i got a question, i need to create a group extension, to make calls to 6 sw phones, but i need to know if asterisk can do help me to get a unique number and check what extension has received less calls than the others, and pass the new call. We got a call center and want to know if we can distribute the calls depending in what extension is available and from the extensions that are available pass the call to the operator that has answered less calls, can i do this with *? can i get statistics from the use for an extension? can anybody help me?? it sounds like you're wanting to use asterisk's call queueing capabilities. look at http://www.voip-info.org/wiki-Asterisk+call+queues for more info. Especially look at the Strategies section on that page, which has a fewestcalls strategy, which basically rings the extension which has taken the fewest calls to date. flynn ___ 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] Strange Crash
On Sun, 2005-01-30 at 03:11 -0600, Steven Critchfield wrote: On Sun, 2005-01-30 at 12:31 +0330, Paradise Dove wrote: hi, just got an strange crash, and don't know what could cause this type of crashs - hardware failure - memory - cpu ? i have 1xTE405P installed with 4xTA750. using fresh kernel 2.6.9 (no patch). * version is latest CVS HEAD. thanks Program terminated with signal 11, Segmentation fault. Cannot access memory at address 0xb80014bc Seg faults can be faulty memory, overheated CPU, but usually it is an error in programming. #0 0xb7fbbce4 in ?? () (gdb) bt #0 0xb7fbbce4 in ?? () #1 0x080d425d in _IO_stdin_used () #2 0x in ?? () Next time provide the asterisk binary along with the core file to gdb so you can get symbol names and line numbers. I always though sig11 was a memory error... eg, faulty memory. At least, when compiling on a machine with bad memory, I always got sig11's in different/random places sometimes it would compile, and then crash later too :) I'd suggest you try and get around an hour to boot memtest, and see how it goes. (From another thread, this is one very nice reason to have a gentoo CD, it comes with bootable memtest. I wish debian would do that too)... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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] TDM400P specs clarification
Hello, I need some clarification on TDM400P. While browsing the Digium site I see that the TDM400P wildcard is being used as base card for other TDM series cards. I wanted to know what basically the TDM400P card offers. If you see the specs it says : The Wildcard TDM400P is a half-length PCI 2.2 compliant card that supports from one to four telephone interfaces for connecting analog telephones or analog lines to a PC The term : one to four telephone interfaces for connecting analog telephones or analog lines In terms of FXO and FXS what does it mean. I can see that it has four RJ 11 sockets. How will you decide which of the four interface to use for what. I mean FXO or FXS. Thanks in advance. Varun ___ 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] Eicon Diva audio problem [Newbie]
Hi
all,
I have trouble
getting my setup configured properly.
I have a Eicon|DIVA
Server BRI-2M/-2F card installed, using melware driver and following asterisk
wiki guidelines.
However whe I try to
dialup the number I get only silence and after a while
disconnection.
The following is
displayed on the console.
What am I doing
wrong ?
*CLI == Starting
CAPI[contr1/0991]/0 at demo,0991,1 failed so falling back to exten
's' -- Executing Wait("CAPI[contr1/0991]/0", "1") in new
stack -- started pbx on channel (callgroup=2)!
== Spawn extension (demo, s, 1) exited non-zero on
'CAPI[contr1/0991]/0'
Thanks and
Kind Regards
Nic
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Re: [Asterisk-Users] Asterisk on MS Virtual Server
On Sun, 2005-01-30 at 14:02 +, Paul Tyreman wrote: Hi, This might not be a very popular question, but I was just wondering if anyone have ever tried to run Asterisk on a Windows computer using Microsoft Virtual Server (http://www.microsoft.com/windowsserversystem/virtualserver/default.mspx). I am told that you can run Linux on a virtual server using this software, so in theory it should be possible. I run a windows based domain, but am also keep run Asterisk. I could use a different computer, but that would use more electricity. So I was thinking on upgrading my server to the latest specs, and trying to run Asterisk on Windows 2003 using this software. If anyone has managed this, I would love to hear from you. Thanks, Paul. Maybe this is not something you would like to consider, but here is *my* 0.02c worth (please read to the end to get the full picture): * Dump windows, and install linux * Install vmware under linux * Install your windows 2003 inside vmware * Install asterisk under linux native * Make sure you start asterisk with -p and as user root This should (in theory) give you a decent asterisk system, and keep your fully functional (well, the same as what you have now) windows server. If your windows domain authentication request is delayed by 90ms, will your users notice? No. If your telephone conversation (audio) is delayed by random times between 10 and 200ms, will your caller's notice? Maybe... Some people report 'acceptable' telephone conversations under pretty bad conditions, but I would prefer the domain authentication stuff to get delayed rather than my call with a customer... Regards, Adam ___ 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 x NAT
Hi All, I have a question for you: - "SIP doesn't work behind NAT very well" Do you agree with this sentence? regards César ___ 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] HDLC for Dummies?
Can any give me or point me to a short and simple explanation of what HDLC is? ___ 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 x NAT
Hi All, I have a question for you: - SIP doesn't work behind NAT very well Do you agree with this sentence? regards César ___ 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 x NAT
César Davi Ávila do Nascimento wrote: Hi All, I have a question for you: - SIP doesn't work behind NAT very well Do you agree with this sentence? Depends on the NAT/firewall in question, you can also alleviate some of these issues using STUN and sip proxing... -- Best regards, Duane http://www.cacert.org - Free Security Certificates http://www.nodedb.com - Think globally, network locally http://www.sydneywireless.com - Telecommunications Freedom http://happysnapper.com.au - Sell your photos over the net! http://e164.org - Using Enum.164 to interconnect asterisk servers In the long run the pessimist may be proved right, but the optimist has a better time on the trip. ___ 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
AW: [Asterisk-Users] HDLC for Dummies?
http://en.wikipedia.org/wiki/HDLC -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Eric Bishop Gesendet: Montag, 31. Januar 2005 11:40 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: [Asterisk-Users] HDLC for Dummies? Can any give me or point me to a short and simple explanation of what HDLC is? ___ 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
RE: [Asterisk-Users] Asterisk with Grandstream ringback
http://fm.grandstream.com/gs/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Robert Rozman Sent: Wednesday, January 26, 2005 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk with Grandstream ringback - Original Message - From: Kim Lux [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 26, 2005 8:13 AM Subject: Re: [Asterisk-Users] Asterisk with Grandstream ringback I updated to firmware version x.22 and this and a few other problems were fixed. I was running x.18 and it allowed me to do a successful upgrade via http. Hi, could you please post your settings for http upgrade and url for firmware ? Thanks, Rob. On Tue, 2005-01-25 at 08:10 -0700, Kim Lux wrote: Are you saying that you are running firmware X.22 and it is not doing the callback when you hang up ? Where exactly did you get that firmware version ? Thanks On Tue, 2005-01-25 at 16:55 +0200, Doug Reid - Stormcorp wrote: Hi All Has any one tested Ver X.22 on the grandstreams? If so have you noticed the problem experienced with ringback? When you hang up the GS rings again and its the same call you put down. Only seen this with Ver X.16 and X.18 not yet with X.22 but I'm still not 100% convinced. Doug ___ 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 -- Kim Lux, Diesel Research Inc. ___ 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
Re: [Asterisk-Users] PRI for Data and Voice
On Mon, 31 Jan 2005, Eric Bishop wrote: Do you have a config sample on how to handle digital PPP sessions in Asterisk? No, but there may be examples in the wiki: http://www.voip-info.org/tiki-pagehistory.php?page=Asterisk+zapraspreview=3 http://www.digium.com/downloads/ppp.txt http://www.digium.com/downloads/hdlc.txt I think the last two are for permanent leased connections and possibly not what you are looking for. Peter ___ 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] Trying to make but it fails
chan_zap.c:3669: dereferencing pointer to incomplete type chan_zap.c:3670: confused by earlier errors, bailing out make[1]: *** [chan_zap.o] Error 2 make[1]: Leaving directory `/usr/src/asterisk/asterisk-1.0.5/channels' make: *** [subdirs] Error 1 [EMAIL PROTECTED] asterisk-1.0.5]# It's not the last errors which are important, but the first. B Thanks for the advice, After a little chase for the top error message i found that i had an old libpri. Per ___ 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] TDM400P specs clarification
On Mon, 2005-01-31 at 15:30 +0500, [EMAIL PROTECTED] wrote: Hello, I need some clarification on TDM400P. The TDM400P card by itself has no use. You purchase a mix of FXS and FXO daughter cards (they are coloured Red and Green) which pug into four available positions on the card. That decides the functionality of the TDM400 card. In terms of FXO and FXS what does it mean. I can see that it has four RJ 11 sockets. How will you decide which of the four interface to use for what. I mean FXO or FXS. -- . . ___. .__ Posix Systems - Sth Africa. e.164 VOIP ready /| /| / /__ [EMAIL PROTECTED] - Mark J Elkins, Cisco CCIE / |/ |ARK \_/ /__ LKINS Tel: +27 12 807 0590 Cell: +27 82 601 0496 ___ 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] x100p issues + TDM400P
Hello, I have been wanting to use digium x100p to get started. But it seems to have compatibility issues for different regions. I am refering to the 600 ohm US pstn standard only. I am in India so If I were not to use x100p card then what card I need to go in for? I also read that x100p has been discontinued. I am being recommended TDM400P wildcard. Is it OK for India. Does TDM400P wildcard has FXO and FXS ? According to the Silicon Labs spec sheet, the chip set on the tdm card includes parameters for India (370 ohm). It should work. ___ 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] Japan
Leo: Do you have a working MGCP call agent config? I've been struggling with such a config for months and all my email queries have gone unanswered. If you have such a config, and even better also have SMDI support configured for/on Asterisk, I'd really appreciate a copy. Thanks,Steve Leo Ann Boon wrote: Jason Frisch wrote: I asked Softbank and it seems that using SIP etc directly is not an option. Something to do with theVoIP-TA being used for communications between the providers call-agent. Sounds like they're using MGCP. At this point, Asterisk is not able to act as an MGCP endpoint, it can only be a 'call agent'. ___ 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 -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-8001 fax: 215-898-9348 sip:[EMAIL PROTECTED] ___ 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] NAT and SIP
So this leads me to believe there's some kind of limit per IP on NAT'ed SIP clients. Can anybody shed some light on this? It sounds like a nat box issue and probably related to port mapping. I've seen the same kind of issue with multiple vpn clients trying to pass through a single nat box. Swapping the box for a different model fixed the problem. The only way to tell for sure is to trace the packets inside and outside the nat box to see exactly what the box is doing. For example, the first sip session will use udp 5060, but on weird nat boxes the second sip session will get mapped to udp 5061 (or something like that), and obviously * isn't listening on that port. ___ 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] Group Extension
i need to create a group extension, to make calls to 6 sw phones, but i need to know if asterisk can do help me to get a unique number and check what extension has received less calls than the others, and pass the new call. We got a call center and want to know if we can distribute the calls depending in what extension is available and from the extensions that are available pass the call to the operator that has answered less calls, can i do this with *? can i get statistics from the use for an extension? can anybody help me?? You're looking for ACD (automatic call distribution). Check the wiki for help: http://voip-info.org/tiki-index.php?page=Asterisk%20config%20queues.conf http://voip-info.org/tiki-index.php?page=Asterisk%20config%20agents.conf ___ 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] NAT and SIP
Hi, Does Asterisk have a limit to how many NAT'ed SIP clients it supports behind a single IP? [...] Theoretical limit is around 65536 clients. But the practical limit is something far less. ___ 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] Caller ID in AU
Nathan If you want more specific information for AUS, drop me a direct mail. My Sipura 3000 passes the PSTN call (on hook) to the asterisk box and also the CLIDNUM. My only problem is that the asterisk box then sends the caller-id to the handset connected to the sipura, I can get the username but the number never shows up even though I can see it in the asterisk messagesthats still soemthing I need to sort Pete -- Open WebMail Project (http://openwebmail.org) -- Original Message --- From: Nathan Alberti [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Mon, 31 Jan 2005 09:54:13 -0500 Subject: Re: [Asterisk-Users] Caller ID in AU I have updated the Wiki with this info as I have seen it come up a few times. Nathan. Gary wrote: Don't forget Howard, that Caller-ID presentation is an extra chargeable service. has it been turned on on these lines and confirmed ?? (its handy to carry a caller-id in your kit for checking:-) On Sat, 29 Jan 2005 07:30:07 +1100, Howard Lowndes wrote: On Fri, 2005-01-28 at 19:02, Simon Brown wrote: Insert a Wait(2) before Answer OK, I'll try that. I have also done the suggested mod to the chan_zap.c module to make the default rings 2. Simon Brown -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Howard Lowndes Sent: Friday, 28 January 2005 17:30 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Caller ID in AU Is anyone in AU successfully getting Caller ID from the analogue PSTN service? If so, what settings? -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 -- Howard. LANNet Computing Associates; Your Linux people http://www.lannetlinux.com -- When you just want a system that works, you choose Linux; when you want a system that just works, you choose Microsoft. -- Flatter government, not fatter government; Get rid of the Australian states. ___ 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 --- End of Original Message --- ___ 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] reason 24 (Call ended with Q.931 cause)
Hi, Enable the driver tracing (see wrapTrace* and libTrace* in oh323.conf), re-run and send me the output file. Michael. Tola Ogunsan wrote: Hi Michael and Everyone I'm trying to connect Asterisk to a CISCO AS5350 using oh323 and I'm getting this error reason 24 (Call ended with Q.931 cause) I've checked the Asterisk wiki and several other resources. Please can anyone give me a hint on what the problem is I reach my wits end. Thanks Tola my config and debug Configuration of OpenH323 channel driver -- Version: 0.7.1 Listening on address: 0.0.0.0:1720 Gatekeeper used: No gatekeeper FastStart/H245Tunnelling/H245inSetup: OFF/OFF/OFF Supported formats in pref. order: g7290 Jitter buffer limits (min/max): 20-500 ms TCP port range: 1 - 2 UDP (RAS) port range: 1 - 2 UDP (RTP) port range: 1 - 2 IP Type-of-Service value: 0 User input mode: 3 Max number of inbound H.323 calls: 10 Max number of outbound H.323 calls: 10 Max number of simultaneous H.323 calls: 10 Max call rate (ingress direction): 1.00/30 Starting simple switch on 'Zap/3-1' -- Executing Wait(Zap/3-1, 1) in new stack -- Executing Dial(Zap/3-1, OH323/[EMAIL PROTECTED]|10) in new stack -- H.323 call to [EMAIL PROTECTED] with codec(s) g729 Outbound H.323 call 'ip$localhost/263'. -- Called [EMAIL PROTECTED] Call 'ip$localhost/263' cleared. -- H.323 call 'ip$localhost/263' cleared, reason 24 (Call ended with Q.931 cause) Call 'ip$localhost/263' cleared in INIT state. -- OH323/L263 is busy -- Hungup 'OH323/L263' == Everyone is busy/congested at this time (1:1/0/0) -- Executing Hangup(Zap/3-1, ) in new stack == Spawn extension (incoming, s, 3) exited non-zero on 'Zap/3-1' -- Hungup 'Zap/3-1' Call 'ip$localhost/263' without owner has already been cleared (2). -- Starting simple switch on 'Zap/3-1' ___ 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
Re: [Asterisk-Users] TDM400P specs clarification
Hello, I need some clarification on TDM400P. While browsing the Digium site I see that the TDM400P wildcard is being used as base card for other TDM series cards. I wanted to know what basically the TDM400P card offers. If you see the specs it says : The Wildcard TDM400P is a half-length PCI 2.2 compliant card that supports from one to four telephone interfaces for connecting analog telephones or analog lines to a PC The term : one to four telephone interfaces for connecting analog telephones or analog lines In terms of FXO and FXS what does it mean. I can see that it has four RJ 11 sockets. How will you decide which of the four interface to use for what. I mean FXO or FXS. The fxo modules are designed to interface to a pstn line (where ringing comes from the central switching office). The fxs modules are designed to have standard telephone sets plugged into them, and the module provides ringing voltage to the telephone. On the digium web site, a tdm04b is the tdm card with four fxo modules installed. The tdm40b is the same tdm card with four fxs modules installed. The tdm22b has two fxo and two fxs modules. If you incorrectly connect a fxs module to a pstn line, you will likely blow the module making it useless. ___ 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 x NAT
I have a question for you: - SIP doesn't work behind NAT very well Do you agree with this sentence? Depends. Asterisk behind a nat box tends to be an implementation problem limited by the knowledge of the person doing the implementation and somewhat by the functionality implemented within the nat box. Sip phones behind a nat box (with asterisk on a registered IP address) tends to be rather easy, and how well it works depends a lot on how well the sip phone vendor implemented nat support. Both asterisk and sip phones behind different nat boxes tends to be the most difficult to implement and requires the greatest amount of knowledge/experience to implement. Again, depends a lot on the functionality provided in the nat boxes. The issue with sip is that session startup and control occurs across udp port 5060, and the two endpoints (* and phone) negotiate another set of udp ports for the rtp (voice) session. The choice of which rtp ports to use was left up to each sip phone vendor, so the udp port number in use could be anything from about 8000 (xlite) to something greater then 32,000. Some firewall/nat boxes will actually watch the sip rtp negotiation process by inspecting the contents of the sip packets, and open up the wanted ports. However, most cheap nat boxes don't do that, and leave it up to you to statically define/map the ports required. ___ 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] ISDN supplemetary services (Hold, Retrieve, 3PTY) on HFC-8S
Hi, I use a Cologne Chip HFC-8S card with chan_capi. (TE mode only) So I've set up mISDN with CAPI and it's working just fine for'normal' calls. But I do need more. Namely Hold, Retrieve and 3PTY which are not supported yet in the mISDN implementation of CAPI, whereas it is in the AVM Fritz Card CAPI driver for instance. So my question is:Is anyone awareof an alternative imlementation that supports these supplementary services? Thanks in advance. Tobias Cermann ___ 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 x NAT
Thanks a lot! Regards César - Original Message - From: Rich Adamson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 31, 2005 9:18 AM Subject: Re: [Asterisk-Users] SIP x NAT I have a question for you: - SIP doesn't work behind NAT very well Do you agree with this sentence? Depends. Asterisk behind a nat box tends to be an implementation problem limited by the knowledge of the person doing the implementation and somewhat by the functionality implemented within the nat box. Sip phones behind a nat box (with asterisk on a registered IP address) tends to be rather easy, and how well it works depends a lot on how well the sip phone vendor implemented nat support. Both asterisk and sip phones behind different nat boxes tends to be the most difficult to implement and requires the greatest amount of knowledge/experience to implement. Again, depends a lot on the functionality provided in the nat boxes. The issue with sip is that session startup and control occurs across udp port 5060, and the two endpoints (* and phone) negotiate another set of udp ports for the rtp (voice) session. The choice of which rtp ports to use was left up to each sip phone vendor, so the udp port number in use could be anything from about 8000 (xlite) to something greater then 32,000. Some firewall/nat boxes will actually watch the sip rtp negotiation process by inspecting the contents of the sip packets, and open up the wanted ports. However, most cheap nat boxes don't do that, and leave it up to you to statically define/map the ports required. ___ 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
Re: [Asterisk-Users] Fwd and Tollfree
Yups... at least via FWD it is still working. Rene Kluwen Chimit - Original Message - From: Liaan vd Merwe To: asterisk-users@lists.digium.com Sent: Friday, January 28, 2005 4:48 PM Subject: [Asterisk-Users] Fwd and Tollfree Hallo all do any of you know if the toll free access to the Netherlands is still working via FWD or Iaxtel? thanks liaan Do you Yahoo!?Yahoo! Search presents - Jib Jab's 'Second Term' ___Asterisk-Users mailing listAsterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo 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
Re: [Asterisk-Users] detailed asterisk howto
szj wrote: Hi, all: I am a newbie to the asterisk and its architecture. :( After reading some help in the tarball of Asterisk, I am still in the mess. So I want to know where I can find a detailed explanation of the Asterisk which including the Architecture, Install, Configure, usage example document. Maybe what I want is too much, after all it is a open project, not commercial product. If I want to get that, will I buy it or take participate in some course to learn that ??? Try this: http://www.asteriskdocs.org/ It walks you through the basic setup info and is pretty well written. Good luck and welcome, Robert Jackson ___ 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] congestion problem with only one number
Michiel van Baak wrote: All is working great. Till I called the bank. It rings one time and then gives me the congestion tone. Here is what I see on the CLI (phone nr obfuscated for privacy reasons): -- Executing Dial(SCCP/michiel-0004, Modem/g1:xx|50|Ttr) in new stack If you want to use your bank's IVR then you will have to remove the t option. If you don't want the fake ring remove the r option. Don't use options you don't understand. ___ 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] TDM400P specs clarification
[EMAIL PROTECTED] wrote: Hello, I need some clarification on TDM400P. While browsing the Digium site I see that the TDM400P wildcard is being used as base card for other TDM series cards. I wanted to know what basically the TDM400P card offers. If you see the specs it says : The Wildcard TDM400P is a half-length PCI 2.2 compliant card that supports from one to four telephone interfaces for connecting analog telephones or analog lines to a PC The term : one to four telephone interfaces for connecting analog telephones or analog lines In terms of FXO and FXS what does it mean. I can see that it has four RJ 11 sockets. How will you decide which of the four interface to use for what. I mean FXO or FXS. You purchase the FXO, FXS, or any combination of FXO and FXS modules. ___ 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 x NAT
I have a question for you: - SIP doesn't work behind NAT very well Do you agree with this sentence? Complete and utter crap (if you assume a few things). SIP w/NAT works just fine if: Asterisk itself is not behind NAT You do not want to use SIP reinvites You use some form of NAT Keepalive* You use nat=yes in sip.conf Your NAT router is not SIP aware If your NAT router is SIP aware then you can 1) turn off it's SIP awareness and treat it like a dumb NAT router or 2) enable it's SIP awareness and turn off nat=yes in sip.conf. A SIP aware router might make reinvites work of both SIP clients have a SIP aware router. * You can keep your NAT alive by using a registration of 60 seconds on the NAT device, or use qualify=yes in sip.conf, or use the NAT Keepalive features of your SIP device. ___ 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] Trunked IAX or not
You must set trunk=yes in the context of the relevant provider. Not all providers support it. The benefit of trunking grows exponentially with the number of calls in progress. -mark On Jan 31, 2005, at 2:24 AM, Spencer Nassar wrote: The test results that Philipp pointed out show some protocol comparisons that include iax2 trunking / alaw and iax2 / alaw and concludes that IAX2 trunking is more than twice as fast as non trunking IAX. Forgive the newbie question, but what is this distinction? In what cases is a connection 'trunking' or 'not'? If I have a register = statement in my iax.conf file, is that a trunked connection to my DiD provider? Thanks! -- Mark Eissler, [EMAIL PROTECTED] Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.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] NAT and SIP
Rich Adamson wrote: For example, the first sip session will use udp 5060, but on weird nat boxes the second sip session will get mapped to udp 5061 (or something like that), and obviously * isn't listening on that port. The port that shows up in sip show peers is the remote SOURCE port and addresss. Asterisk does not normally care about such things. Any NAT router that modified the DESTINATION port and address would not not work. Does anyone know of a basic NAT for Dummies document that I can point people to? This is something that comes up again and again from people that don't understand how NAT works. ___ 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] Strange Crash
memtest86 is a nice tool and if you go to their site(http://memtest86.com), they have an ISO bootable image there also. Lyle - Original Message - From: Adam Goryachev [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 31, 2005 4:26 AM Subject: Re: [Asterisk-Users] Strange Crash On Sun, 2005-01-30 at 03:11 -0600, Steven Critchfield wrote: On Sun, 2005-01-30 at 12:31 +0330, Paradise Dove wrote: hi, just got an strange crash, and don't know what could cause this type of crashs - hardware failure - memory - cpu ? i have 1xTE405P installed with 4xTA750. using fresh kernel 2.6.9 (no patch). * version is latest CVS HEAD. thanks Program terminated with signal 11, Segmentation fault. Cannot access memory at address 0xb80014bc Seg faults can be faulty memory, overheated CPU, but usually it is an error in programming. #0 0xb7fbbce4 in ?? () (gdb) bt #0 0xb7fbbce4 in ?? () #1 0x080d425d in _IO_stdin_used () #2 0x in ?? () Next time provide the asterisk binary along with the core file to gdb so you can get symbol names and line numbers. I always though sig11 was a memory error... eg, faulty memory. At least, when compiling on a machine with bad memory, I always got sig11's in different/random places sometimes it would compile, and then crash later too :) I'd suggest you try and get around an hour to boot memtest, and see how it goes. (From another thread, this is one very nice reason to have a gentoo CD, it comes with bootable memtest. I wish debian would do that too)... Regards, Adam -- -- Adam Goryachev Website Managers Ph: +61 2 8304 [EMAIL PROTECTED] Fax: +61 2 9345 4396www.websitemanagers.com.au ___ 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
Re: [Asterisk-Users] Record inbound and outbound calls to and from one number.
Tim Mattison wrote: Good call. For our American readers... does anyone know where I can obtain a list of states/counties and their regulations in regards to call recording? I would think the Public Utilities Commission for each state, but that's just a guess. A quick tickle of google came up with: http://www.rcfp.org/taping/ * I take no responsibility for their content. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] congestion problem with only one number
On 07:42, Mon 31 Jan 05, Eric Wieling wrote: Michiel van Baak wrote: All is working great. Till I called the bank. It rings one time and then gives me the congestion tone. Here is what I see on the CLI (phone nr obfuscated for privacy reasons): -- Executing Dial(SCCP/michiel-0004, Modem/g1:xx|50|Ttr) in new stack If you want to use your bank's IVR then you will have to remove the t option. If you don't want the fake ring remove the r option. Don't use options you don't understand. Even without any options I get the same result: -- Executing Dial(SCCP/michiel-000b, Modem/g1:xx|50) in new stack -- Modem[i4l]/ttyI3 is busy -- Hungup 'Modem[i4l]/ttyI3' == Everyone is busy/congested at this time When I call my cell the second after that all works fine. The 2 ISDN lines are only connected to the * box, so no other hardware can claim the line. Michiel ___ 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] Asterisk@home and Zap Channels
I have one more question that I can't seem to get straight, The ZAP channel phone, I can't dial any other extentions from it, I just get a fast busy. Same if I dial 9 to use the outside trunk. It works great from the SIP soft phone, but I can't seem to get the FXS phone to behave. In your zapata.conf, where you defined your FXS port, you have to put it in the right context so it as access to the other extensions. Just put the same as your softphone ex.: extension=localstations HTH ___ 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] Error while trying to execute asterisk
asterisk -cv Jan 31 18:03:20 WARNING[13145]: cdr_addon_mysql.c:264 my_load_module: Unable to load config for mysql CDR's: cdr_mysql.conf [app_addon_sql_mysql.so] = (Simple Mysql Interface) [pbx_dundi.so]Jan 31 18:03:20 WARNING[13145]: loader.c:258 ast_load_resource: /usr/lib/asterisk/modules/pbx_dundi.so: undefined symbol: pbx_substitute_variables_varshead Jan 31 18:03:20 WARNING[13145]: loader.c:440 load_modules: Loading module pbx_dundi.so failed! - what is the problem with asterisk __ Do you Yahoo!? The all-new My Yahoo! - Get yours free! http://my.yahoo.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] Announcement to caller when called party has picked up - without initial Answer()?
This is super easy to do. All you need to do is to put that announcement in a MP3 and set a musiconhold class for that incoming Zap channel. Then basically when ever that PSTN number rings, Asterisk will play the MP3 stream Your call may be monitored or recorded, please hangup if you do not agree...etc in a loop until the line is answered. Caller doesn't pay a single dime to listen to your announcement(s). So now you have announcement on ringing! David Liu Hong Kong On Mon, 31 Jan 2005 15:09:48 +0100, Stefan Gofferje wrote Hi folks, is there a chance to play an announcement to the calling party AFTER the called party has picked up the receiver and WITHOUT asterisk answering the call? I have a special line where conversations should be recorded on. German federal laws forbid recording without consent, so the idea is to play an announcement like This line is been monitored, please hang up if you don't agree. Trouble is, I don't want asterisk to answer the call and therefore produce cost for the caller because it may be that there is nobody present to answer the call. Asterisk should just send ring indications to the PSTN and play the announcement when the call is picked up. Any ideas? Regards, Stefan -- (o_ Stefan Gofferje | Linux Systems Specialist //\ Reg'd Linux User #247167 | SuSE Certified Linux Trainer V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside ___ 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
Re: [Asterisk-Users] Asterisk with Grandstream ringback
That URL has been locked down for resellers and vendors only for a couple of days now. Pity, one of the good things about the Grandstream was their freely available firmwares. Oh well, time to find another phone - the Sipura 841 is looking interesting. Craig - Original Message - From: Doug Reid - Stormcorp [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Monday, January 31, 2005 7:05 PM Subject: RE: [Asterisk-Users] Asterisk with Grandstream ringback http://fm.grandstream.com/gs/ -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Robert Rozman Sent: Wednesday, January 26, 2005 10:01 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk with Grandstream ringback - Original Message - From: Kim Lux [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Wednesday, January 26, 2005 8:13 AM Subject: Re: [Asterisk-Users] Asterisk with Grandstream ringback I updated to firmware version x.22 and this and a few other problems were fixed. I was running x.18 and it allowed me to do a successful upgrade via http. Hi, could you please post your settings for http upgrade and url for firmware ? Thanks, Rob. On Tue, 2005-01-25 at 08:10 -0700, Kim Lux wrote: Are you saying that you are running firmware X.22 and it is not doing the callback when you hang up ? Where exactly did you get that firmware version ? Thanks On Tue, 2005-01-25 at 16:55 +0200, Doug Reid - Stormcorp wrote: Hi All Has any one tested Ver X.22 on the grandstreams? If so have you noticed the problem experienced with ringback? When you hang up the GS rings again and its the same call you put down. Only seen this with Ver X.16 and X.18 not yet with X.22 but I'm still not 100% convinced. Doug ___ 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 -- Kim Lux, Diesel Research Inc. ___ 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
Re: [Asterisk-Users] congestion problem with only one number
Even without any options I get the same result: -- Executing Dial(SCCP/michiel-000b, Modem/g1:xx|50) in new stack -- Modem[i4l]/ttyI3 is busy -- Hungup 'Modem[i4l]/ttyI3' == Everyone is busy/congested at this time When I call my cell the second after that all works fine. The 2 ISDN lines are only connected to the * box, so no other hardware can claim the line. That is specific to I4L and I can't help with that, other than to point out that not many people use I4L with Asterisk. They susually use CAPI or ZapBRI drivers. ___ 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] Announcement to caller when called party has picked up - without initial Answer()?
What you want is impossible! How can you expect Asterisk to play a message to the caller without answering the phone? On Mon, 31 Jan 2005, Stefan Gofferje wrote: Hi folks, is there a chance to play an announcement to the calling party AFTER the called party has picked up the receiver and WITHOUT asterisk answering the call? I have a special line where conversations should be recorded on. German federal laws forbid recording without consent, so the idea is to play an announcement like This line is been monitored, please hang up if you don't agree. Trouble is, I don't want asterisk to answer the call and therefore produce cost for the caller because it may be that there is nobody present to answer the call. Asterisk should just send ring indications to the PSTN and play the announcement when the call is picked up. Any ideas? Regards, Stefan ___ 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] x100p issues + TDM400P
Does TDM400P wildcard has FXO and FXS ? This card as 4 ports, and on each port you can put an FXS or FXO module. So you can make any combination : 2 FXO and 2 FXS, 4 FXS, 4 FXO, etc HTH ___ 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] Announcement to caller when called party haspicked up - without initial Answer()?
-Original Message- From: David Liu [mailto:[EMAIL PROTECTED] Sent: 31 January 2005 14:34 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Announcement to caller when called party haspicked up - without initial Answer()? This is super easy to do. All you need to do is to put that announcement in a MP3 and set a musiconhold class for that incoming Zap channel. Then basically when ever that PSTN number rings, Asterisk will play the MP3 stream Your call may be monitored or recorded, please hangup if you do not agree...etc in a loop until the line is answered. Caller doesn't pay a single dime to listen to your announcement(s). So now you have announcement on ringing! David Liu Hong Kong To play music on hold first the call would need to be answered!?!?!? Its straight forward to play an announcement to a caller just prior to dialing an extension. However what you need to do is establish whether a channel is infact busy before issuing the Answer() command. I suspect that this is either not possible or would require some huge dirty hack. But someone more knowledgable than myself could tell you am sure. That aside, TBH I think as a customer I would prefer to pay the couple of pence to wait on hold (knowingly) / be asked to leave a message / be told I am X in queue / etc than to only hear ringing. After about 20seconds of ringing I would give up. Being put on hold I would be more forgiving. And whats more important to you, customers that give up and look elsewhere or customers that have to pay for 30-60seconds of holding IF they choose to. Also I was under the impression that in Europe calls are charged as soon as you start ringing and not on pickup (this may be out of date as its been a while since my school skiing trip ;-P ) HTH alex This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. ___ 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] Announcement to caller when called party has picked up - without initial Answer()?
Hello Stefan, Am Mo, den 31.01.2005 schrieb Stefan Gofferje um 15:09: is there a chance to play an announcement to the calling party AFTER the called party has picked up the receiver and WITHOUT asterisk answering the call? I would try the M option in the Dial-Command. See http://www.voip-info.org/wiki-Asterisk+cmd+dial Best regards Kai -- Kai Militzer WESTEND GmbH | Internet-Business-Provider Technik CISCO Systems Partner - Authorized Reseller Lütticher Straße 10 Tel 0241/701333-11 [EMAIL PROTECTED] D-52064 Aachen Fax 0241/911879 ___ 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] Announcement to caller when called party has picked up - without initial Answer()?
Remco Barende wrote: What you want is impossible! How can you expect Asterisk to play a message to the caller without answering the phone? One-way audio before answer is a pretty standard telco feature with PRI service in some parts of the world. ___ 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] Trunked IAX or not
On January 31, 2005 08:57 am, Mark Eissler wrote: You must set trunk=yes in the context of the relevant provider. Not all providers support it. The benefit of trunking grows exponentially with the number of calls in progress. Isn't it just a linear savings? 1 call: UDP overhead + voice data 2 calls: UDP overhead + voice data + voice data 3 calls: UDP overhead + 3xvoice data etc... without trunking the UDP overhead is repeated for each voice call -A. ___ 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] Announcement to caller when called party has picked up - without initial Answer()?
On Mon, 31 Jan 2005, Remco Barende wrote: What you want is impossible! How can you expect Asterisk to play a message to the caller without answering the phone? It can be done on isdn connection and over VoIP links as well. The reverse audio path is (can be) opened before the answer. The answer allows the forward path to be opened. E.g. you can use Playback(someFile|noanswer) to play a custom busy message without answering the line on isdn. Dial() application will answer the incoming line once it is ready to bridge the two calls together. If nothing else then one can always modify the Dial() application to play a specific sound just prior to sending the answer. I have not checked if there already is a generic way to hook into Dial that early. Peter ___ 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] Audio Quality over LAN very bad
Hi All, I'm running Asterisk on the following vendor_id : GenuineIntelmodel name : Celeron (Coppermine)cpu MHz : 668.202cache size : 128 KB with 192 MB Ram Audio coming from Asterisk (the demo ) is excellent when using a SIP phone on the LAN to Asterisk, and when dialling in from outside via ISDN to Asterisk. However, when connecting from SIP phone to SIP phone (across LAN) and dialling from externally to SIPwhich is on the local LAN it is very choppy and one can barely make out the other party. I'm using an Eicon Diva 2-m card and 100mb network all round. What could be the cause as I believe bandwidth is ruled out. Thanks and regards Nic ___ 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] Announcement to caller when called party haspicked up - without initial Answer()?
On Mon, 31 Jan 2005, Alex Barnes wrote: Also I was under the impression that in Europe calls are charged as soon as you start ringing and not on pickup (this may be out of date as its been a while since my school skiing trip ;-P ) Not in all countries at least. Sweden has always had the charge start only on answer. I expect most of Europe to use a similar convention. At least I have never encountered a payphone in Europe that consumed my coins before the line was answered. Peter ___ 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] Asterisk with Grandstream ringback
On Monday 31 January 2005 14:28, Craig Guy wrote: That URL has been locked down for resellers and vendors only for a couple of days now. Pity, one of the good things about the Grandstream was their freely available firmwares. Oh well, time to find another phone - the Sipura 841 is looking interesting. [... 5 stupid signatures removed. Why are people so lazy and ignorant? ...] You can still get Grandstream firmware, just not possibly broken bleeding edge. B ___ 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] how to stop ringing after congestion.
From what I have read and understood, the
On Mon, 31 Jan 2005, el Flynn wrote:
Jon Gabrielson wrote:
When there are no zap channels available, I signal congestion
using the following:
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Playtones(congestion)
exten = _9NXX,3,Congestion
The congestion sound plays correctly, but the ringing continues
in the background. Why is it still ringing and how do I make it stop?
try
exten = _9NXX,3,Congestion(5)
which will stop the tones after 5 seconds.
flynn
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RE: [Asterisk-Users] Trunked IAX or not
Isn't it just a linear savings? 1 call: UDP overhead + voice data 2 calls: UDP overhead + voice data + voice data 3 calls: UDP overhead + 3xvoice data etc... without trunking the UDP overhead is repeated for each voice call I know nothing about the IAX protocol but I wont let that stop me from offering opinion! :-D Could it be because RTP requires another connection for control (RTPC). I think is one port higher than the data port!?!?!?! IAX has a saving for this to maybe? If that's wrong then I stand corrected but that's my laymans understanding. alex This email and any attached files are confidential and copyright protected. If you are not the addressee, any dissemination, distribution or copying of this communication is strictly prohibited. Unless otherwise expressly agreed in writing, nothing stated in this communication shall be legally binding. ___ 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] Callgroup with bristuff ISDN?
Remco Barende wrote: Hi list! I'm still trying to figure out about the groups in asterisk. If I understand correctly, if you assign a certain group number and you assign the same call group number to a sip device the device will reing even though you did not specifically specify it in extension.conf? Can I use callgroups in such a setup, any config examples? Which isdn channel are you using ? Chan Capi, Zaphfc, mIsdn, isdn4l ?? ___ 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] Callgroup with bristuff ISDN?
Remco Barende ha scritto: Hi list! I'm still trying to figure out about the groups in asterisk. If I understand correctly, if you assign a certain group number and you assign the same call group number to a sip device the device will reing even though you did not specifically specify it in extension.conf? How will this work for ISDN BRI/PRI? I don't want some extensions to get all calls from the BRI/PRI, just the calls from one DID. The wiki gives an example whereby a callgroup= is linked to a channel but this seems kinda silly with ISDN. Can I use callgroups in such a setup, any config examples? Thanks!! ___ 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
Re: [Asterisk-Users] Announcement to caller when called party haspicked up - without initial Answer()?
On Monday 31 January 2005 14:37, Alex Barnes wrote: [...] Also I was under the impression that in Europe calls are charged as soon as you start ringing and not on pickup (this may be out of date as its been a while since my school skiing trip ;-P ) I'm not sure that has ever been the case. B ___ 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] Strange Crash
memtest86 is a nice tool and if you go to their site(http://memtest86.com), they have an ISO bootable image there also. Knoppix also can be used to test memory On the boot prompt just type memtest and it will start the test HTH ___ 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] Announcement to caller when called party has picked up - without initial Answer()?
Peter Svensson wrote: Dial() application will answer the incoming line once it is ready to bridge the two calls together. If nothing else then one can always modify the Dial() application to play a specific sound just prior to sending the answer. I have not checked if there already is a generic way to hook into Dial that early. I looked at show application dial when I read the question earlier today. There is a hook for playing a message to the recipient of the Dial before patching them together, but I didn't see anything for the other way around. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/ ___ 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] Re: grandstream budgetone-100 updates
dean collins wrote: Im using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.2025 Jan 18:25 Error Aborted 192.168.16.32C:\Program Files\TFTP Desktop\1.0.5.18\cfg.txt Octet, Send192.168.16.2025 Jan 18:25 Error Can anyone tell me why these fail each time? Probably because the files are not in your TFTP root. This is probably because you are not using these files to autoconfigure your phones. Stephen R. Besch ___ 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] how to stop ringing after congestion.
My apologies - Dan
On Mon, 31 Jan 2005, Dan Adams wrote:
From what I have read and understood, the
On Mon, 31 Jan 2005, el Flynn wrote:
Jon Gabrielson wrote:
When there are no zap channels available, I signal congestion
using the following:
exten = _9NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _9NXX,2,Playtones(congestion)
exten = _9NXX,3,Congestion
The congestion sound plays correctly, but the ringing continues
in the background. Why is it still ringing and how do I make it stop?
try
exten = _9NXX,3,Congestion(5)
which will stop the tones after 5 seconds.
flynn
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Re: [Asterisk-Users] Callgroup with bristuff ISDN?
If I understand correctly, if you assign a certain group number and you assign the same call group number to a sip device the device will reing even though you did not specifically specify it in extension.conf? Can I use callgroups in such a setup, any config examples? Which isdn channel are you using ? Chan Capi, Zaphfc, mIsdn, isdn4l ?? zaphfc (bristuff) ___ 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] music on hold that starts at beginning of file
Hi, Id like to have asterisk play a sound file while a caller is waiting to be connected to an extension. I tried using music on hold, but that seems to run in a loop, not playing from the beginning for each caller. Are there any other options? It doesnt have to be an MP3 file. I tried sending a background() command before the dial(), but that doesnt appear to work. Thanks in advance - Joe ___ 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] Announcement to caller when called party has picked up - without initial Answer()?
How do you want to play something on the line without answering it first ? ___ 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] Tuning MoH Volume
I'm using * 1.0.3 on Gentoo 2004.3, zaprtc from bri-stuff for timing. When I put a caller on hold, the volume of the hold music in the callers ear is extremely loud. I'm using the default entry from the musiconhold.conf: default = quietmp3:/var/lib/asterisk/mohmp3 Volumes with a called or calling party are fine, it's just the hold music volume that seems to be way off kilter. Anyone know if it's possible to do any fine tuning of the volume? ___ 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] detailed asterisk howto
still in the mess. So I want to know where I can find a detailed explanation of the Asterisk which including the Architecture, Install, Configure, usage example document. The answers to the questions you've been asking are probably here: Starter articles: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html Full install etc. http://automated.it/guidetoasterisk.htm And of course: http://www.asteriskdocs.org ___ 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 x NAT
I'll agree with that sentence. There are many times when even STUN and so on isn't going to help. In Guatemala, a lot of people end up with private IPs, behind two NATs, etc. I've seen them aggressively timeout connections, limit the range of ports available for NAT (to a ridiculously low number), etc. etc. We gave up on SIP and are now using IAX for our customer phones. -Michael -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of César Davi Ávila do Nascimento Sent: Monday, January 31, 2005 5:56 AM To: Asterisk-Users@lists.digium.com Subject: [Asterisk-Users] SIP x NAT Hi All, I have a question for you: - SIP doesn't work behind NAT very well Do you agree with this sentence? regards César ___ 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] SPA-841 Call Waiting
Am I doing something wrong here? GOt a SPC-841 the other day and have it registering properly. Can place and recive calls as expected but when on the phone, a second call is immediately dumped to busy voicemail. Does this thing not support call-waiting? Or, have I just got my configs wrong? Paul -- Paul A. DugasDugas Enterprises, LLC [EMAIL PROTECTED]1711 Indian Ridge Drive p:404-932-1355 f:770-516-4841 Woodstock, GA 30189-6856 USA ___ 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] TDM400P specs clarification
On Mon, 31 Jan 2005 06:12:53 -0600, Rich Adamson [EMAIL PROTECTED] wrote: If you incorrectly connect a fxs module to a pstn line, you will likely blow the module making it useless. Wow, i think it shoud be more fool-proof ;) Lucky I tried first with an analog phone in my TDM11B... Julian. ___ 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] Asterisk and Cisco phones chan_sccp vs chan_skinny vs native SIP and one-way audio
Gents, I've recently built a couple of Asterisk boxes and want to migrate away from CallManager to Asterisk. On my Asterisk box I have about 8 Grandstream BT101s and a Cisco 7905G in SIP mode, on my CallManager I have about 10 x 30VIP, 2 x 7940 and a 7960. I've built Asterisk version 1.0.5 along with Zozo's chan_sccp (CVS latest from last night) and got it partially working. All devices are on the inside of a private network at the moment (192.168.144.0/24) and I'm having some issues with devices on chan_sccp. The 30VIPs can place and receive calls but I have a one-way audio problem. The 7960 can receive calls but when I place calls from it I end up directly in the voicemail unavailable and the SIP phone doesn't ring. Looking at the network the SIP device opens an RTP stream to the Cisco (30VIP or 7960) but the Cisco device doesn't send RTP back to the SIP phone... can anyone point me in the right direction with this? A more general question: with Cisco phones being removed from a CallManager environment, is it best to keep them in Skinny/SCCP mode or change out to SIP? The 30VIPs can only do SCCP/Skinny so which of the two channel drivers in Asterisk should I use for best effect? Mike ___ 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] AGI Processing Order
Apparently I have had a few calls show up in my logs as something odd happening. Apparently at a certain spot the wrong number of digits are being presented, but I am not sure why that is. That is what I am trying to figure out. I was curious, does anyone know of a wiki page that outlines the order the different AGI files are processed in? Dan ___ 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] Re: grandstream budgetone-100 updates
Nope the files are there. Extracted the entire zip file into the same folder. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Stephen R. Besch Sent: Monday, January 31, 2005 10:17 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Re: grandstream budgetone-100 updates dean collins wrote: I'm using tftp server that automatically loads on each reboot, for some reason the last 2 files fail to load each time. (and I think this has always been the case) Aborted 192.168.16.32C:\Program Files\TFTP Desktop\1.0.5.18\cfg000b82005c24 Octet, Send 192.168.16.2025 Jan 18:25 Error Aborted 192.168.16.32C:\Program Files\TFTP Desktop\1.0.5.18\cfg.txt Octet, Send192.168.16.2025 Jan 18:25 Error Can anyone tell me why these fail each time? Probably because the files are not in your TFTP root. This is probably because you are not using these files to autoconfigure your phones. Stephen R. Besch ___ 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] Linksys RT31P2-NA
I am noticing a problem with the RT31P2-NA when it loses internet. Has anyone experienced problems where it does not reconnect to asterisk and obtain its dialtone again? Brian Fertig Planet Telecom, Inc. ___ 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] Sending forwarded calls out to a different provider
Hi, Is it possible to send calls that are forwarded from a Cisco 7900 phone using the Call Forward All feature out using a different service provider or group like another SIP trunk? I don't want to tie up our incoming lines that are ZAP so I was thinking about getting a secondary service for just forwarded calls. Thanks, Calvin -- S i x F e e t U p | Nowhere to go but open-source Silicon Valley: +1 (650) 401-8579 | Midwest: +1 (317) 861-5948 Toll-Free: 1-866-SIX-FEET mailto:[EMAIL PROTECTED] http://www.sixfeetup.com | Zope Hosting from $19.95/month ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Announcement to caller when called party haspicked up - without initial Answer()?
Stefan if I understood what you need, maybe this works...
MSG_FILE=/var/spool/asterisk/
exten = s,1,Dial(SIP/MyPhone|60|M(playmessage^${MSG_FILE}))
[macro-playmessage]
exten = s,1,Wait(0.5)
exten = s,2,Playback(${ARG1})
exten = s,3,SetVar(MACRO_RESULT=CONTINUE)
I didn´t try it but I think this should work...
pls let me know if it did in fact work... (if u want u can do it off list)
bye,
M.
- Original Message -
From: Stefan Gofferje [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, January 31, 2005 11:45 AM
Subject: Re: [Asterisk-Users] Announcement to caller when called party
haspicked up - without initial Answer()?
Remco Barende schrieb:
What you want is impossible!
How can you expect Asterisk to play a message to the caller without
answering the phone?
You got me wrong... Asterisk should answer the call not initially but
when the called party picked up.
Incoming call
|
|
send ring indicator
|
|
called party picks up receiver --- yes answer, announce, bridge
|
no
|
Congestion on timeout
--
(o_ Stefan Gofferje | Linux Systems Specialist
//\ Reg'd Linux User #247167 | SuSE Certified Linux Trainer
V_/_ Linux is like a Wigwam - No gates, no windows, Apache inside
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RE: [Asterisk-Users] Call Waiting Audio Prompt
On Mon, 2005-01-31 at 09:37 +, Alex Barnes wrote: Thanks for the replies everyone how do you expect to get the indication that you have a callwaiting call? The whole point is I don't want it. The beep is a guard that hides the caller-id fsk spill also. So you can't get callwaiting-callerid and not have a beep. I don't really need that either, users can see whos waiting on hold for them via the web page so anything caller waiting related is fine but only as long as it doesn't have negative impact on the call quality. Setting the indication durations to zero has helped hugely as the sound is far less intrusive now. Also all of the end points are essentially SIP phones (or DECT phones plugged into 2102's) so callerID doesn't need to be passed inbound. I will try what Jon sugggests: Zaptel.conf callwaiting=no callwaitingcallerid=no It didn't really occur to me as was looking at configuring the SIP side. Then you need to do the tricks in SIP to not send more than one call at a time to the endpoint. The changes in zaptel.conf won't help here. -- Steven Critchfield [EMAIL PROTECTED] ___ 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
