[Asterisk-Users] I thought they weren't charging - FW: [DIDx.net] Happy holidays wishes from DIDX.net.
Did anyone else get this? I thought they weren't charging? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 28, 2005 9:35 PM To: x Subject: [DIDx.net] Happy holidays wishes from DIDX.net. Dear x, The DIDX.net team wishes you a very happy holiday season. DIDX has revised its monthly rates' structure. We will no longer charge you anything to be a Regular Member in the DIDX network. Once you are comfortable with DIDX and are ready to start your trading on the DIDxchange, you will be required to keep a minimum of 20 DID's buy or sell total. This is a Regular Membership. Otherwise, you will be charged a minimum monthly fee of $20. You can avoid this charge by purchasing 20 DID's for as low as 10 cents each. This will total $22 a month for 20 DID's with our commission charges. Thank you for joining and being a part of the successful DIDX network, the fastest growing VOIP exchange in the world. * To un-subscribe to our news letter, Please login to your account, click on edit my info, and you can unscubscribe to this news letter from there. You can not un-subscribe to our notification emails.. RefFile: DIDx - Email.pm ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] select codec based on extension
Leandro Rzezak ha scritto:
I'm having same problem. Were you able to solve it?
No, codecs became a secondary problem later in our project so we ended
up with 711 on all servers and more bandwidth, anyway the post refers
to asterisk 1.0.something and I never investigated the problem in more
detail... I think it's possible, usually when you receive no answers (as
the case of that post) you have made a really silly question :)
On 10/18/05, *Simone Cittadini* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I've the following installation :
|asterisk client| --- |asterisk server| --- |other asterisk
server|
all the connections are made in IAX, the client and first server
allows
711 and 729
the other server only allows 729 since it has low bandwidth at
disposal
all the numbers but a few are routed to a digium card in the first
server, the others are routed to the other server, this way :
[default]
exten = _123X.,1,Dial(IAX2/otherserver/${EXTEN})
exten = _123X.,2,Hangup
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Hangup
when I call 123456 from the client box ...
on the client :
Call accepted by asterisk server (format alaw)
on the server :
Call accepted by other asterisk server (format g729)
on the other server :
Called [EMAIL PROTECTED]
and then on the server in the middle :
Oct 18 18:00:37 NOTICE[2846]: channel.c:1724 ast_set_write_format:
Unable to find a path from alaw to g729
Oct 18 18:00:37 NOTICE[2846]: channel.c:1757 ast_set_read_format:
Unable
to find a path from g729 to alaw
since that something at the end of the call and the paps which sits
before the first asterisk server both have g729, I don't like too
much
having to pay to translate something which need not translation.
Is there a clever combination of sip.conf, iax.conf and
extensions.conf
I'm missing to solve my problem ?
___
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RE: [Asterisk-Users] Bad Checksum answering inbound call
The server is hosted in Souls data center and the connection is using SIP to a VEP directly on their voip backbone. We have tried putting a Sip phone directly on to the connection and it works fine. Thanks Darren From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Thursday, 29 December 2005 5:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Bad Checksum answering inbound call ummm - can we have some more detail? ISDN? BRI? PRI? Analog? PaulH Blackburn Melb - Original Message - From: Darren Younger To: asterisk-users@lists.digium.com Sent: Wednesday, December 28, 2005 8:09 PM Subject: [Asterisk-Users] Bad Checksum answering inbound call Could anybody please help me with problem.. Outbound calls work fine, however inbound calls ring the phone, then answering the call, the service provider doesnt receive the picked up message from asterisk. We have narrowed it down to an incorrect checksum in the packets being sent back from asterisk after answering an inbound call. Regards, Darren Younger National Solutions Architect Nightfire Technologies Pty Ltd [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM2400 wierdness
Kerry: I hope this helps. I had EXACTLY the same symptom when I was trying to get an X100P clone to work yesterday. Bumping the toneduration parameter in zapata.conf to 200 milliseconds cured the problem. Roger Kerry Garrison wrote: Asterisk 1.2.1 Updated the TDM2400 driver over the weekend Incoming calls seem to work perfectly Outbound calls never connect. If you listen in on the call to a 7 digit local number, you hear the first 6 digits, then a small delay, then the last digit. Then there is a long pause before the line is picked up, then a very long pause before the telco fires back you call could not be completed at this time. Calling using an analog phone on that line works fine. Do I possibly have some DTMF issues or something like that? Any suggestions would be appreciated. This is my only installation with the TDM2400 so I am kind of at a loss. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Bad Checksum answering inbound call
OK - what does the screen on the Asterisk box say? Off hand, it sounds like a codec issue. But the fact that your service provider says that you are not providing an answer signal points more to a dialplan. (ie: Adding an 'answer' to the dialplan) PaulH - Original Message - From: Darren Younger To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, December 29, 2005 8:08 PM Subject: RE: [Asterisk-Users] Bad Checksum answering inbound call The server is hosted in Souls data center and the connection is using SIP to a VEP directly on their voip backbone. We have tried putting a Sip phone directly on to the connection and it works fine. Thanks Darren From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: Thursday, 29 December 2005 5:42 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Bad Checksum answering inbound call ummm - can we have some more detail? ISDN? BRI? PRI? Analog? PaulH Blackburn Melb - Original Message - From: Darren Younger To: asterisk-users@lists.digium.com Sent: Wednesday, December 28, 2005 8:09 PM Subject: [Asterisk-Users] Bad Checksum answering inbound call Could anybody please help me with problem.. Outbound calls work fine, however inbound calls ring the phone, then answering the call, the service provider doesnt receive the picked up message from asterisk. We have narrowed it down to an incorrect checksum in the packets being sent back from asterisk after answering an inbound call. Regards, Darren Younger National Solutions ArchitectNightfire Technologies Pty Ltd [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] select codec based on extension
Off hand, I agree that it's probably doable...even if you have to put
another sip server inbetween.
(or pay the $10 per channel for the g729 licence if it's only a few
channels)
PaulH
- Original Message -
From: Simone Cittadini [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, December 29, 2005 7:52 PM
Subject: Re: [Asterisk-Users] select codec based on extension
Leandro Rzezak ha scritto:
I'm having same problem. Were you able to solve it?
No, codecs became a secondary problem later in our project so we ended
up with 711 on all servers and more bandwidth, anyway the post refers
to asterisk 1.0.something and I never investigated the problem in more
detail... I think it's possible, usually when you receive no answers (as
the case of that post) you have made a really silly question :)
On 10/18/05, *Simone Cittadini* [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
I've the following installation :
|asterisk client| --- |asterisk server| --- |other asterisk
server|
all the connections are made in IAX, the client and first server
allows
711 and 729
the other server only allows 729 since it has low bandwidth at
disposal
all the numbers but a few are routed to a digium card in the first
server, the others are routed to the other server, this way :
[default]
exten = _123X.,1,Dial(IAX2/otherserver/${EXTEN})
exten = _123X.,2,Hangup
exten = _X.,1,Dial(Zap/g1/${EXTEN})
exten = _X.,2,Hangup
when I call 123456 from the client box ...
on the client :
Call accepted by asterisk server (format alaw)
on the server :
Call accepted by other asterisk server (format g729)
on the other server :
Called [EMAIL PROTECTED]
and then on the server in the middle :
Oct 18 18:00:37 NOTICE[2846]: channel.c:1724 ast_set_write_format:
Unable to find a path from alaw to g729
Oct 18 18:00:37 NOTICE[2846]: channel.c:1757 ast_set_read_format:
Unable
to find a path from g729 to alaw
since that something at the end of the call and the paps which
sits
before the first asterisk server both have g729, I don't like too
much
having to pay to translate something which need not translation.
Is there a clever combination of sip.conf, iax.conf and
extensions.conf
I'm missing to solve my problem ?
___
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Re: [Asterisk-Users] Bad Checksum answering inbound call
OK - what does the screen on the Asterisk box say? - Original Message - From: Darren Younger To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Sent: Thursday, December 29, 2005 8:08 PM Subject: RE: [Asterisk-Users] Bad Checksum answering inbound call The server is hosted in Souls data center and the connection is using SIP to a VEP directly on their voip backbone. We have tried putting a Sip phone directly on to the connection and it works fine. Thanks Darren From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]Sent: Thursday, 29 December 2005 5:42 PMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re: [Asterisk-Users] Bad Checksum answering inbound call ummm - can we have some more detail? ISDN? BRI? PRI? Analog? PaulH Blackburn Melb - Original Message - From: Darren Younger To: asterisk-users@lists.digium.com Sent: Wednesday, December 28, 2005 8:09 PM Subject: [Asterisk-Users] Bad Checksum answering inbound call Could anybody please help me with problem.. Outbound calls work fine, however inbound calls ring the phone, then answering the call, the service provider doesnt receive the picked up message from asterisk. We have narrowed it down to an incorrect checksum in the packets being sent back from asterisk after answering an inbound call. Regards, Darren Younger National Solutions ArchitectNightfire Technologies Pty Ltd [EMAIL PROTECTED] ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Problems with multiple outbound calls going to PSTN - Wildcard TE405P
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm having an outbound calling issue with our SIP phones. When one call is made to the PSTN another person trying to call receives a 404 error on the SIP phone. If we call the PSTN using SIP phone A and also calling from SIP phone B to SIP phone C everything works. The only problem we're seeing is multiple calls going to the PSTN. Please let me know if anyone has any suggestions or recommendations. I haven't use PRI line, and from your explanation I guess that you need to define that every next calls needs to use next free line. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] select codec based on extension
On Thu, December 29, 2005 9:52, Simone Cittadini said: Leandro Rzezak ha scritto: I'm having same problem. Were you able to solve it? No, codecs became a secondary problem later in our project so we ended up with 711 on all servers and more bandwidth, anyway the post refers to asterisk 1.0.something and I never investigated the problem in more detail... I think it's possible, usually when you receive no answers (as the case of that post) you have made a really silly question :) Either that or noone really knows the answer... ;-) -- F Peeters PIII 450 - 1 GB - * 1.2 - BRIstuff 0.3.0 Pre 1 - Florz patch 2 Sweex HFC-PCI modes=2 sync_slave=2 timer_card=0 Cologne HFC-S pins #52, #54, #55 connected in parallel for synching. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call test
Hello, My server was offline when you tried to call. try again please What are your problems Regards Harry --- aturntablist [EMAIL PROTECTED] a écrit : didnt work for me, but having problems -- http://www.unitedhearts.co.uk/ - Meet someone special, make new friends. http://www.qwertyhosting.net/ - Advanced hosting services. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] call test
Your SER is requesting authentication from my SIP client. On 28/12/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello, I need to test my configuration please to dial sip:[EMAIL PROTECTED] . Your call will be sent to a queue . Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Peter Bowyer Email: [EMAIL PROTECTED] Tel: +44 1296 768003 VoIP: sip:[EMAIL PROTECTED] VoIP: [EMAIL PROTECTED] FWD: **275*5048707000 VoipTalk: **473*5048707000 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Problems with multiple outbound calls going toPSTN - Wildcard TE405P
- Original Message - From: Tomislav Parcina [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, December 29, 2005 8:58 PM Subject: [Asterisk-Users] Re: Problems with multiple outbound calls going toPSTN - Wildcard TE405P In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... I'm having an outbound calling issue with our SIP phones. When one call is made to the PSTN another person trying to call receives a 404 error on the SIP phone. If we call the PSTN using SIP phone A and also calling from SIP phone B to SIP phone C everything works. The only problem we're seeing is multiple calls going to the PSTN. Please let me know if anyone has any suggestions or recommendations. I haven't use PRI line, and from your explanation I guess that you need to define that every next calls needs to use next free line. er - no. With PRI, the system will automaticaly try to use free lines. No need to define this. PaulH ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What is the best Dell Machine for Asterisk?
On 29 Dec 2005, at 01:02, William Boehlke wrote: The 830s are nice but limited because they do RAID on a card and have but one suitable PCI slot. So you can have an interface card or RAID, but not both. That's not true. I just built a system on a Dell 830 with the RAID card. There are three PCI slots in total and one of them fit the Eicon DIVA Server card I'm using. I've been using Dell rackmounts at work for years now and never had any issues. This is the first time I went for a tower server, there is no rack at home... The box isn't entirely noise-free, but compared to the equivalent rackmount models it is very quiet, you could call it pantry-friendly as opposed to living room-friendly. The machine runs CentOS 4.2 (RHEL 4.2 with the VIN numbers scraped off) and Asterisk 1.2.1 with chan_capi. Granted, I'm not a heavy user, but I like the voice quality I'm getting. jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] s or _X. , is there any change since Asterisk 1.2
Hi, Before Asterisk 1.2 release, s extension never worked for my sip phone and I had to catch calls in my [incoming] using _X. but today after installing Asterisk 1.2, extension s is doing the same thing what _X. used to do. My understanding was that s extension was good only for FXO. Is there any change in its behavior since 1.2 that it is treating calls incoming on sip same as incoming on FXO. If so, which extension should be used and why? I am totally confused. Thanks Zeeshan A Zakaria ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP! Asterisk 1.2.1 stops immediately - voicemail problem?
Hi, our production system stops immediately when a caller hangs up without leaving a voicemail. This is the last output from the console: -- Playing 'vm-isunavail' (language 'de') -- Playing 'vm-intro' (language 'de') -- Playing 'beep' (language 'de') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/1189/INBOX/msg format: wav49, 0x821ba18 -- x=1, open writing: /var/spool/asterisk/voicemail/default/1189/INBOX/msg format: gsm, 0x823a0f0 -- x=2, open writing: /var/spool/asterisk/voicemail/default/1189/INBOX/msg format: wav, 0x823a3d8 -- User hung up What can be wrong? Thanks for any help!! BK ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI: This number has been disconnected
On Wed, 2005-12-28 at 14:00 -0300, Javier Ergas wrote: I believe this behavior has nothing to do with the [EMAIL PROTECTED] Scripts. I think the problem is in the PRI signalization. I can see the zap hangup messages when trying to call a disconnected number. . -- Executing Dial(SIP/9349-1787, ZAP/g0/2514990) in new stack -- Called g0/2514990 -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' == No one is available to answer at this time -- Executing Goto(SIP/9349-1787, s-NOANSWER|1) in new stack -- Goto (macro-dialout-trunk,s-NOANSWER,1) The telco says they are sending inband information with the status of the call, but Asterisk is hanging up the channel instead of connecting it to let hear the audio message. There is a post with a similar issue here: http://mailgate.supereva.com/comp/comp.dcom.isdn.capi/msg04138.html Is anyone experiencing the same behavior? Sounds like the difference between doing inband signalling or out of band signalling. I think by default, a PRI uses out of band signalling, ie, it just sends a message saying this number if un reachable so asterisk just hangs up and plays the local congestion dialplan. What you need to do is use inband signalling, so that asterisk won't hangup, and instead will pass the audio from the telco through. See /etc/asterisk/zapata.conf: ; PRI Out of band indications. ; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to work ; outofband: Signal Busy/Congestion out of band with RELEASE/DISCONNECT ; inband: Signal Busy/Congestion using in-band tones priindication = outofband Regards, Adam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Congestion problem
I use to have this in my extensions.conf [fxo4] exten = s,1,Dial(SIP/214,20,tw) exten = s,2,Answer exten = s,3,VoiceMail,u214 exten = s,4,Congestion exten = s,5,Hangup exten = s,102,Answer exten = s,103,VoiceMail,b214 exten = s,104,Congestion exten = s,105,Hangup When somebody calls me on fxo4 port * sents that call to SIP 214 phone. The problem is that when call ends and SIP user hangs up, the line stays up. Now I don't use Congestion any more. Can sombody tell me do I realy need that congestion signal? On http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+Congestion they say that congestion waits that other party hangs up. Why would I wait for that? Please, any informations are welcome. -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP to SIP calls
Hello, nobody use an ip phone on these mailing lists ! your call will put on queue . I just need some people to dial sip:[EMAIL PROTECTED] to check and debug my config . Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Conditional CODEC translation
Dear Leandro, I do not think you can avoid the translation here, You must have the licences to be able to talk to your provider OR you may want to allow your ip phone to use g729 I do not think there is any other way. You can try the intel's trial version to see if that works for you, at no cost to you, however its not for commercial usage. Rehan We have a VoIP termination provider that allows g729. We would that internal calls (between our own IP phones) be handled using alaw, and outgoing calls using native forwarded g729 without translation (ie, not using asterisk g729 licenses). We need to avoid translations. WHAT WE HAVE NOW: IP Phone --alaw-- IP Phone IP Phone --alaw-- Asterisk --g729-- VoIP provider (Phones are configured only to allow alaw and g729, provider is configured only to allow g729; however phones are never using g729) WHAT WE NEED: IP Phone --alaw-- IP Phone IP Phone --g729-- VoIP provider Please help me accomplish that. Thank you -- Leandro Rzezak [EMAIL PROTECTED] Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip debug file.txt
Tzafrir Cohen wrote: On Wed, Dec 28, 2005 at 04:22:27PM -0500, Leonard Burton wrote: Greetings, How can I log the output of sip debug into a file? Obviously, # asterisk -rx sip debug debug.txt did not produce the desired results logger.conf ? I usually do asterisk -rvn | tee /tmp/sipdebug.txt Then turn on sip debug on the cli. This captures everything. You need to make sure that the debug output is sent to the console in logger.conf /Olle ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Driver not configuring correctly on TE210P forCCS
It was jumpers, thanks for the reply. Sorry for the spam everyone. It would be helpful to get a little quick start card with these cards to save me a day of head aches but not to worry. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Liew Sent: 28 December 2005 22:45 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Driver not configuring correctly on TE210P forCCS Hi Alex, Have you checked that your jumper setting on the card has been shorted for E1. Its open by default to T1 - which overrides your zaptel.conf settings. Cheers, Paul Information contained in this e-mail and any attachments are intended for the use of the addressee only, and may contain confidential information of Ubiquity Software Corporation. All unauthorized use, disclosure or distribution is strictly prohibited. If you are not the addressee, please notify the sender immediately and destroy all copies of this email. Unless otherwise expressly agreed in writing signed by an officer of Ubiquity Software Corporation, nothing in this communication shall be deemed to be legally binding. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SIP to SIP calls
Dear Harry, What would you like to be debugged ? Is nxs.yi.org your server ? Rehan Hello, nobody use an ip phone on these mailing lists ! your call will put on queue . I just need some people to dial sip:[EMAIL PROTECTED] to check and debug my config . Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Go directly to new messages from VoiceMailMain?
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
I want to create an extension that goes directly to my new messages
without having to press 1. How do I do that? I can call
VoiceMailMain but then I have to choose 1 from the menu. I'd like it
to go there and play the first message or say There are no new
messages and hangup. How can I do this?
exten = 298,1,Ringing
exten = 298,2,Wait(2)
exten = 298,3,VoiceMailMain(s${CALLERIDNUM}) ; if pass is the same lik
extension number
--
Tomislav Parcina
[EMAIL PROTECTED]
___
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[Asterisk-Users] Re: Grandstream Configuration Utility available
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... FYI - Grandstream has made available for public download, a config tool for the Budgtone and GXP phones, as well as the Handytone adapters. It is available for download here. http://www.grandstream.com/y-configurationtool.htm What exacly do you configure with it? -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] I thought they weren't charging - FW: [DIDx.net] Happy holidays wishes from DIDX.net.
Dear Don, We are still not charging. The BASIC Membership is free, however if you choose to be an active member, you can do so by 1. Either Be Active by trading on the exchange. 2. Pay a monthly minimum charge Linda Did anyone else get this? I thought they weren't charging? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Wednesday, December 28, 2005 9:35 PM To: x Subject: [DIDx.net] Happy holidays wishes from DIDX.net. Dear x, The DIDX.net team wishes you a very happy holiday season. DIDX has revised its monthly rates' structure. We will no longer charge you anything to be a Regular Member in the DIDX network. Once you are comfortable with DIDX and are ready to start your trading on the DIDxchange, you will be required to keep a minimum of 20 DID's buy or sell total. This is a Regular Membership. Otherwise, you will be charged a minimum monthly fee of $20. You can avoid this charge by purchasing 20 DID's for as low as 10 cents each. This will total $22 a month for 20 DID's with our commission charges. Thank you for joining and being a part of the successful DIDX network, the fastest growing VOIP exchange in the world. * To un-subscribe to our news letter, Please login to your account, click on edit my info, and you can unscubscribe to this news letter from there. You can not un-subscribe to our notification emails.. RefFile: DIDx - Email.pm ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: What is the best Dell Machine for Asterisk?
I am ordering a Dell 2800 today. From averything I have read, there should be no issues with the box. It will be SMP and with the following changes I have made my 1750, very stable. REF: FYI on zttool output on SMP system --- Results after 56 passes --- Best: 100.00 -- Worst: 99.987793 -- Average: 99.999564 Only 2 were 99.987793, the 54 others were all 100.00. I got this by making the changes below on my dual proc Dell 1750. setpci -v -s 01:08.1 LATENCY_TIMER=8 setpci -v -s 00:0f.1 LATENCY_TIMER=8 setpci -v -s 01:04.0 LATENCY_TIMER=8 setpci -v -s 01:02.0 LATENCY_TIMER=8 setpci -v -s 00:0f.2 LATENCY_TIMER=8 setpci -v -s 01:04.0 LATENCY_TIMER=8 (these are USB, SCSI HW RAID driver, Ethernet, Video, etc. I did not alter ZAP cards, nor any bridges or buses) echo 1 /proc/irq/17/smp_affinity (Ethernet) echo 1 /proc/irq/18/smp_affinity (SCSI HW RAID Driver) echo 2 /proc/irq/20/smp_affinity (TDM) echo 2 /proc/irq/24/smp_affinity (TE411P) I also turned of the startup of irqbalance. The setpci changes did the most work concerning reaching 100% in zttest. Irqbalance was causing the the processor handling the interrupts of the zap cards to change very often. This would impose a delay during the change and cause the zttest numbers to drop/be inconsistent. Because I turned irqbalance off, the irqs are processed round robin style, which is also not good. Therefore, I hard coded the processor affinity for the zap cards to one proc and all other high load irqs to the other proc. If you have more than 2 procs, you can spread them out even more. If you do not turn off irqbalance, the affinity changes will be overwritten by it. I made these changes on a live system without issue. I set these changes in /etc/rc.d/rc.local to reset them after reboots. -- -- Steven May you have the peace and freedom that come from abandoning all hope of having a better past. ---- --- - - - -- - - -- - - - --- - -- - - --- - - -- - -- -- - -- Phil Pritchard [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] i could not recommend dell more. and also to say, there is loads of support for 2.4 and 2.6 kernels for most if not all of there hardware. times are a changing! the best bang for buck(only a few months ago) is the poweredge sc430..(extremely quiet, without a load) works perfectly, everything! with regard to irq problems, you will get that with any machine/hardware. not just dell. when i build or commission a new server, i always run hardware tests. like ram, hard drive, processor and motherboard. checking motherboard for interrupt conflicts etc.(with all cards installed) the best all round tool for this is the UBCD. universal boot cd ( goggle ) there are a few versions, ie. a small(about 35meg) and a large one( abit over 100meg)...for memory. i have a lot of dell machines, including some very old ones.. that still go hard... the engineering in these machines is the reason why i use them. i have pulled a lot of machines/servers apart, and there one of the best.. there my bit! hope it helps... Phil William Boehlke wrote: We have more than a hundred Dell servers in production at customers. We use them because we can have them serviced easily, just about anywhere. They are principally 1850s and 2850s, or their predecessors, in T1 and larger applications. The reported IRQ problems are easily avoided if, for example, you don't use more than one card in a server or, better yet, don't use cards at all. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of C F Sent: Tuesday, December 20, 2005 5:22 PM To: Walt Reed; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] What is the best Dell Machine for Asterisk? Last time I checked Dell does support Linux. http://configure.us.dell.com/dellstore/config.aspx?c=uscs=04kc=6W300l=en; oc=pe2850-maxs=bsd Scroll down to the OS chose and you'll see out of 26 choices only 4 are M$, the rest minus one are for a total of 21 linux choices. On 12/19/05, Walt Reed [EMAIL PROTECTED] wrote: Why oh why would you want to install *, which runs on Linux, on a machine made by a company that does NOT support Linux? Both IBM and HP do a pretty good job of supporting Linux. So do other Linux oriented companies like PenguinComputing.com Digium cards have historically been a little finicky in regards to which machines they work in, but Sangoma cards should work in virtually any modern machine that has the right type of slots (careful with some modern servers that ONLY have PCI Express slots.) Hopefully someone can comment about modern digium cards in regards to compatability. Have they gotten better? On Mon, Dec 19, 2005 at 08:44:38AM +0800, Hiu Yen Onn said: Then, how about Acer? Does it work well with asterisk? Simone Cittadini wrote: Matt Florell ha scritto: The best Dell for a production environment Asterisk server is no Dell at all.
[Asterisk-Users] Re: Re: Problems with multiple outbound calls going toPSTN - Wildcard TE405P
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... er - no. With PRI, the system will automaticaly try to use free lines. No need to define this. Well, then it has to be something else :)) -- Tomislav Parcina [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spandsp fax
OK Carlos, Let me know if you need any assistance. Rehan We re trying to use 1.2.1. Im finishing compiling a new version, as soon as I got it Ill let you know. Thanks a lot for the intention. Regards, Carlos From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Rehan Ahmed Sent: Wednesday, December 28, 2005 7:43 PM To: Dov Bigio; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] spandsp fax Which version of Asterisk are you using ? 1.2 had problems in Make file for me 1.0.9 worked with a charm. You can email me with the error you have, maybe I can help you Rehan On 12/28/05, Dov Bigio [EMAIL PROTECTED] wrote: I am using Red Hat 9, but I don't think this changes the procedure - Original Message - From: Carlos Alperin [EMAIL PROTECTED] To: 'Dov Bigio' [EMAIL PROTECTED]; 'Asterisk Users Mailing List -Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 8:24 PM Subject: RE: [Asterisk-Users] spandsp fax Don, The previous question I believe was what linux are you using? By the way, I would like to know that too, just I was trying to make this work for weeks with no success. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dov Bigio Sent: Tuesday, December 27, 2005 10:54 AM To: Kristof Hardy; Asterisk Users Mailing List - Non-CommercialDiscussion Subject: Re: [Asterisk-Users] spandsp fax Hi BJ, Kristof, It worked! I am using the version at http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21c/asterisk-1. 2.x/. I think I had bad symlinks on /usr/local/lib and by reading the tutorial on AsteriskGuru I found that... (The previously installed version of spandsp has been 0.0.3, but now you have installed version 0.0.2. The problem is that the installation of version 0.0.3 creates a symlink, which is not replaced by installation of version 0.0.2. So the symlink points to the library of version 0.0.3, which actually does not exist.). I simply deleted all files related to spandsp from this directory and installed it again! Thank you Dov - Original Message - From: Kristof Hardy [EMAIL PROTECTED] To: Dov Bigio [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-CommercialDiscussion asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 12:59 PM Subject: Re: [Asterisk-Users] spandsp fax Dov Bigio wrote: I am using Asterisk 1.2.1 and followed instructions on http://www.asteriskguru.com/tutorials/spandsp.html to install faxing capability on my server. what platform are you running on? (wich distro?) Does the make of the app_txfax and app_rxfax work out well? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rehan Ahmed AllahWala http://www.SuperTec.com - Tommrow's Technology, Today. http://www.didx.net - DID Number Exchange and Peering Service. Super Technologies Inc., Pensacola, Florida http://www.SuperTec.com - Technologies from tomorrow, Today! ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP! Asterisk 1.2.1 stops immediately - voicemail problem? - SOLVED
we removed the settings for emailbody and emailsubject Kib Eki wrote: Hi, our production system stops immediately when a caller hangs up without leaving a voicemail. This is the last output from the console: -- Playing 'vm-isunavail' (language 'de') -- Playing 'vm-intro' (language 'de') -- Playing 'beep' (language 'de') -- Recording the message -- x=0, open writing: /var/spool/asterisk/voicemail/default/1189/INBOX/msg format: wav49, 0x821ba18 -- x=1, open writing: /var/spool/asterisk/voicemail/default/1189/INBOX/msg format: gsm, 0x823a0f0 -- x=2, open writing: /var/spool/asterisk/voicemail/default/1189/INBOX/msg format: wav, 0x823a3d8 -- User hung up What can be wrong? Thanks for any help!! BK ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM2400 wierdness
Thanks, I will try that. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Hill Sent: Thursday, December 29, 2005 1:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM2400 wierdness Kerry: I hope this helps. I had EXACTLY the same symptom when I was trying to get an X100P clone to work yesterday. Bumping the toneduration parameter in zapata.conf to 200 milliseconds cured the problem. Roger Kerry Garrison wrote: Asterisk 1.2.1 Updated the TDM2400 driver over the weekend Incoming calls seem to work perfectly Outbound calls never connect. If you listen in on the call to a 7 digit local number, you hear the first 6 digits, then a small delay, then the last digit. Then there is a long pause before the line is picked up, then a very long pause before the telco fires back you call could not be completed at this time. Calling using an analog phone on that line works fine. Do I possibly have some DTMF issues or something like that? Any suggestions would be appreciated. This is my only installation with the TDM2400 so I am kind of at a loss. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Conditional CODEC translation
Hello Leandro, Indeed, your problem is a nice one. I do not think this is possible to do this with *. If I am wrong, the list, please correct me... There are two ways of doing that: 1/. would be to have the IP phone have a logic that advertises the preferred codec based on B-number. I do not know of any IP Phones that are able to do that... 2/. would be to have * perform allow/disallow parameters based on the number you have dialed. Both would be interesting... Maybe we will implement this in LoudHush (for the softphone side). Could such a conditional codec be implemented on asterisk in a future version? Bogdan Moldovan MODULO Consulting "The Future Is Not What It Used To Be" http://www.modulo.ro From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Leandro RzezakSent: Thursday, December 29, 2005 6:03 AMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Conditional CODEC translation We have a VoIP termination provider that allows g729.We would that internal calls (between our own IP phones) be handled using alaw, and outgoing calls using native forwarded g729 without translation (ie, not using asterisk g729 licenses). We need to avoid translations.WHAT WE HAVE NOW:IP Phone --alaw-- IP PhoneIP Phone --alaw-- Asterisk --g729-- VoIP provider(Phones are configured only to allow alaw and g729, provider is configured only to allow g729; however phones are never using g729)WHAT WE NEED:IP Phone --alaw-- IP PhoneIP Phone --g729-- VoIP providerPlease help me accomplish that.Thank you-- Leandro Rzezak[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Aastra firmware 1.3.x
I have a related question about the 480i and firmware 1.3... I have a 480i that I got about 1yr. ago and it didn't work well at all. I finally got around to updating the firmware. However, the phone will not load its firmware. I set up the tftp server and I pointed the phone to it. I can watch the logs on the tftp server and see that the transfer initiates. However, at a point in the startup process, the Aastra locks up. The little progress wheel on the display freezes and it won't respond to anything. Not the keypad, Web interface, not even (IIRC) pings. Has anyone run into this before? On Mon, Dec 26, 2005 at 01:02:13PM +0100, BennyBad wrote: Using the: # headset tx gain: # headset sidetone gain: handset tx gain: 10 handset sidetone gain: 0 # handsfree tx gain: 2 Worked great for Me ! Actually we have 10 480i's and the settings are not the same for all phones. handset tx gain xx varies form +5 to +10, to get the same result. So I believe this is a HW issue. Reg. BennyB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert La Ferla Sent: 24. december 2005 04:22 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Aastra firmware 1.3.x (Far-End sound level issue) Taco Scargo wrote: Hello, Just bought two 480i's which I updated to firmware 1.3 I experience the 'Far-End sound level issue' now. I tried configuring the handset tx gain: value but can only make it sound softer, not louder. If there is someone that has managed to get decent Far-end sound level, could he or she please e-mail their used values ? I have a similar issue with the Aastra 9133i and recorded .wav voicemail files. The recorded wav is too soft. I need to find a way to boost the volume level. Does anyone have any solutions or ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users --- [This E-mail scanned for viruses by Declude Virus] -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] SNOM 360 locked up
On Friday 23 December 2005 00:39, Steven Ringwald wrote: On Thu, 2005-12-22 at 23:34 +0100, Christian Stredicke wrote: Try loading http://phone-ip-address/line_sip.htm?settings=saveuser_dp_str1= (if that was in the line 1) while the phone boots up (keep your finger on the reload button). If that does not work, you need to do a tftp update. Yeah. The website address didn't work. (The phone, I think, is not far enough along to even start the webserver). I will try the tftp update method, and see what happens. So far, though, it doesn't seem to be hitting the tftp server that I set up manually. A step by step description can be found here: http://www.snom.com/wiki/index.php/Main_Page#Firmware_Update Also consider moving to version 4.5 (http://www.snom.com/snom360_release_notes.html). Any idea how to do that? I think it is running 4.1. I have put the firmware image URL into the upgrade line before, and it didn't take. (Ended up going back to what it had previously had). Thanks for the help! Steve Regards, Sven ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SPA-3000 + call waiting
Hi, I've got my * machine running, and it's connected to the pstn via a Sipura SPA-3000. My PSTN line has the call waiting feature and I was wondering how * deals with that. All incoming calls are prompted to enter the desired extension, so I was wondering what happens when I'm on the phone ( using the PSTN) and someone calls. I tried it and I found out that I hear the same beep on the phone (just like if I was directly connected to the pstn line) but when I press flash, nothing really happens. I saw that some stuff displayed on the console at this time, so * must have received something. Please let me know if you need more details, Regards, -- Ugo - Please don't send a copy of your reply by e-mail. I read the list. - Please avoid top-posting, long signatures and HTML, and cut the irrelevant parts in your replies. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] smsq
has anyone had any luck compiling and installing the smsq.c utility. I went through the tutorial online and found i was getting errors all the way through it. this is the tutorial i was using... http://www.voip-info.org/wiki-Asterisk+cmd+Sms any light on this subject would be greatly appreciated. begin:vcard fn:Chris Songer n:Songer;Chris org:Blaze Media Inc email;internet:[EMAIL PROTECTED] title:Database Administrator/ IT Director tel;work:615-491-2459 tel;cell:615-491-2459 x-mozilla-html:TRUE url:http://www.getblaze.com version:2.1 end:vcard ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SNOM 360 locked up SOLVED
He had run into a deadlock situation where he entered an (illegal) string for the dial plan that made the phone lock up right after reboot. That bug was fixed in one of the early 4.x versions. The way out was a little trick with the web browser. Generally I think if people have a problem today they should move to 4.5. This version seems to be pretty stable, we did not get any crash-complains or major problem reports from this version. For those who want to move on (feature-wise), it is time to jump on the 5.x train - the 5.0 version has been released a few days ago. We tried our best to test this version as good as possible (including an Asterisk-lab test), but from experience we know that new features always take a certain time to stabilise. Therefore, I would today move to 5.0 only if it has a feature that the 4.5 does not have. We tried to keep the release notes as informative as possible to make this decision as easy as possible for you. Happy New Year! Christian -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Michael George Sent: Thursday, December 29, 2005 10:17 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SNOM 360 locked up SOLVED On Thu, Dec 22, 2005 at 03:58:07PM -0800, Steven Ringwald wrote: Thank you so much for your help, Christian! Your suggestion worked perfectly, and the phones came back up without a problem. What part of his suggestion? Upgrading the firmware to 4.5 via the tftp server? Please elaborate for the benefit of others who may run into this problem. Thank you. -- -M There are 10 kinds of people in this world: Those who can count in binary and those who cannot. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Easiest way to use HFC-S?
What is the easiest way to install and use HFC-S card on Asterisk? As less kernel compiling driver installations as possible. Is it mISDN, or chan_capi, or vISDN, or zaphfc, or? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a Gateway
Nitesh Divecha wrote:
Are there any examples of dial plans? Like how to make the default
context?
I just need a kick start on the config part, as I am really struggling
on routing the calls.
Here is a very very simple example using a PRI. You will need more error
routing in a real dial plan:
extensions.conf:
[general]
static=yes
writeprotect=no
country=us
[local]
include = default
[globals]
TRUNK=Zap/g1
LDTRUNK=Zap/g2
[trunk]
;Long distance pstn
exten = _1NXXNXX,1,Dial(${LDTRUNK}/${EXTEN})
exten = _1NXXNXX,2,Hangup
;pstn
exten = _X.,1,Dial(${TRUNK}/${EXTEN})
exten = _X.,2,Hangup
[default-out]
;This is where you sent trusted calls from sip.conf out to pstn
include = trunk
[default]
;you send incoming pstn calls here as well as untrusted voip calls.
;here you would route call to local numbers you own via enum or static.
exten = 6153247060,1,Wait(2) ; you need to wait
; long enough to get
; CNAM off line
;send incoming call to your register server.
exten = 55,2,Dial(SIP/[EMAIL PROTECTED])
sip.conf:
[general]
bindport = 5060
bindaddr = 0.0.0.0
context = default ; non trusted call from sip side go here
srvlookup = yes
dtmfmode=info
disallow=all
allow=ulaw
allow=alaw
allow=g729
[trusted]
type=friend
context=default-out ; trusted call can go out pstn
host=192.168.0.1
canreinvite=no
zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
span=3,1,0,esf,b8zs
bchan=49-71
dchan=72
span=4,1,0,esf,b8zs
bchan=73-95
dchan=96
loadzone = us
defaultzone=us
zapata.conf:
[channels]
context=default;pstn incoming call go here
switchtype=national
signalling=pri_cpe
toneduration=500
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=-1.0
txgain=-1.0
callerid=asreceived
;
group=1
channel=1-23
channel=73-95
;
group=2
channel=25-47
channel=49-71
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Re: [Asterisk-Users] Asterisk as a Gateway
See the message I post right before this one for a simple example. Ray Yang wrote: Apart from the dial plan issue, can anyone let Asterisk act like Cisco GW to accept SIP call without registered in advance? I've tried this for a long time but no answer yet. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] SPA-3000 + call waiting
You REALLY don't want to have call waiting on a line going into any PBX. You are only asking for problems. My basic home setup is an SPA-3000 but the PSTN line only has call forward on busy, when busy, the number is forwarded to a DID at iax.cc. Kerry Garrison Publisher - http://GeekGazette.com - http://VOIPSpeak.net (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Ugo Bellavance Sent: Thursday, December 29, 2005 6:23 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] SPA-3000 + call waiting Hi, I've got my * machine running, and it's connected to the pstn via a Sipura SPA-3000. My PSTN line has the call waiting feature and I was wondering how * deals with that. All incoming calls are prompted to enter the desired extension, so I was wondering what happens when I'm on the phone ( using the PSTN) and someone calls. I tried it and I found out that I hear the same beep on the phone (just like if I was directly connected to the pstn line) but when I press flash, nothing really happens. I saw that some stuff displayed on the console at this time, so * must have received something. Please let me know if you need more details, Regards, -- Ugo - Please don't send a copy of your reply by e-mail. I read the list. - Please avoid top-posting, long signatures and HTML, and cut the irrelevant parts in your replies. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] smsq
On 29 Dec 2005, at 15:28, chris songer wrote: has anyone had any luck compiling and installing the smsq.c utility. I went through the tutorial online and found i was getting errors all the way through it. this is the tutorial i was using... http://www.voip-info.org/wiki-Asterisk+cmd+Sms any light on this subject would be greatly appreciated. It is part of Asterisk 1.2.1 (that's what I have here, not sure about earlier versions). jens ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] TDM2400 wierdness
Try adding a w in your dial statement. Asterisk will dial even if the line is not ready with a dialtone, adding a w will wait a bit and then dial the number. On 12/29/05, Kerry Garrison [EMAIL PROTECTED] wrote: Thanks, I will try that. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Hill Sent: Thursday, December 29, 2005 1:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM2400 wierdness Kerry: I hope this helps. I had EXACTLY the same symptom when I was trying to get an X100P clone to work yesterday. Bumping the toneduration parameter in zapata.conf to 200 milliseconds cured the problem. Roger Kerry Garrison wrote: Asterisk 1.2.1 Updated the TDM2400 driver over the weekend Incoming calls seem to work perfectly Outbound calls never connect. If you listen in on the call to a 7 digit local number, you hear the first 6 digits, then a small delay, then the last digit. Then there is a long pause before the line is picked up, then a very long pause before the telco fires back you call could not be completed at this time. Calling using an analog phone on that line works fine. Do I possibly have some DTMF issues or something like that? Any suggestions would be appreciated. This is my only installation with the TDM2400 so I am kind of at a loss. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Go directly to new messages from VoiceMailMain?
Tomislav Parcina wrote:
In article [EMAIL PROTECTED], [EMAIL PROTECTED]
says...
I want to create an extension that goes directly to my new messages
without having to press 1. How do I do that? I can call
VoiceMailMain but then I have to choose 1 from the menu. I'd like it
to go there and play the first message or say There are no new
messages and hangup. How can I do this?
exten = 298,1,Ringing
exten = 298,2,Wait(2)
exten = 298,3,VoiceMailMain(s${CALLERIDNUM}) ; if pass is the same lik
extension number
Thanks but that doesn't address my problem. The s option is for the
password. I'm asking about how to get it to play new messages right
away without the user having to press 1
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RE: [Asterisk-Users] Re: Go directly to new messages fromVoiceMailMain?
I believe that there currently is no option for Auto-play
You would have to edit the source code for that.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Robert La Ferla
Sent: Thursday, December 29, 2005 10:49 AM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Go directly to new messages
fromVoiceMailMain?
Tomislav Parcina wrote:
In article [EMAIL PROTECTED],
[EMAIL PROTECTED]
says...
I want to create an extension that goes directly to my new
messages
without having to press 1. How do I do that? I can call
VoiceMailMain but then I have to choose 1 from the menu.
I'd like
it to go there and play the first message or say There are no new
messages and hangup. How can I do this?
exten = 298,1,Ringing
exten = 298,2,Wait(2)
exten = 298,3,VoiceMailMain(s${CALLERIDNUM}) ; if pass is the same
lik extension number
Thanks but that doesn't address my problem. The s option
is for the password. I'm asking about how to get it to play
new messages right away without the user having to press 1
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[Asterisk-Users] Asterisk Server Hangs
Hey guys, Asterisk Server Specs : Asterisk version : CLI show version Asterisk SVN-trunk-r7230 built by [EMAIL PROTECTED] on a i686 running Linux on 2005-12-25 16:14:47 UTC System details : Centos 4.2 (Final) Linux ip-pbx 2.6.9-22.ELsmp #1 SMP Intel Dual Xeon 3.06Ghz Intel SE7501CW2 Motherboard Digium cards : T110P (E1) , TDM22B, TDM31B, TDM24012B I added TDM24012B yes'day but haven't configured or used it yet. Its just connected to the system. The same problem used to occur before adding TDM24012B to the mix. This setup hangs up i,e total freeze cant ssh, cant login even from the system console and nothing in system logs or asterisk logs point me to any obvious problem. There is no coredump in /tmp too. Asterisk also freezes up the server when i issue a stop now command in the CLI sometimes. The only call traffic at this moment are SIP to SIP internal calls, SIP to ZAP external calls and ZAP to SIP incoming calls. In all there must be a total of 10 simultaneous calls. Im using queues, rxfax, txfax, voicemail, meetme (still testing). This happens three or four times in a day. I cant see any IRQ misses in zttool and zttest output is below. Opened pseudo zap interface, measuring accuracy... 99.987793% 99.987793% 99.987793% 99.987793% 100.00% 100.00% 99.987793% 99.987793% 100.00% 100.00% 100.00% 99.987793% 100.00% 99.987793% 100.00% Best: 100.00 -- Worst: 99.987793 -- Average: 99.992300 Found the below messages in dmesg but seems informational rather than a error. Dec 27 22:04:24 asterisk kernel: zaptel Disabled echo canceller because of tone (tx) on channel 32 Dec 29 21:02:12 asterisk kernel: zaptel Disabled echo canceller because of tone (rx) on channel 35 I dont know what the problem could be. I followed the doc at http://www.voip-info.org/wiki-Asterisk+debugging and started asterisk using safe_asterisk and applied the logger related changes. Wat else i can do to debug this issue ? Dushyanth ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] zaptel TDM21B 4-5 second pause
Hi, Sorry if this is a little off topic as its really more zaptel related, but hopefully someone will have come across this., I am noticing a 4-5 second pause when my Digium TDM21B is dialing, just before dialing the last digit. This is causing me problems here in the UK as some telco (no prizes for guessing which one) seems to have reduced thier tolerence on DMTF pauses on some switches, so the switch is timing out after ten digits and not getting the eleventh because of the pause. The installation is [EMAIL PROTECTED] v2.2. an example prefix this is happining on is: +44(0)199255. I have worked around the problem by reducing DTMF_PAUSE in digits.h and recompliing zaptel, but this seems kludgey, does anyone know of a better solution? Many thanks for any help. -Alex. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] What does Page application do?
Why would you use this? Can someone please elaborate on the below description? I'm missing the intent of it. localhost*CLI show application Page [Synopsis] Pages phones [Description] Page(Technology/ResourceTechnology2/Resource2[|options]) Places outbound calls to the given technology / resource and dumps them into a conference bridge as muted participants. The original caller is dumped into the conference as a speaker and the room is destroyed when the original caller leaves. Valid options are: d - full duplex audio q - quiet, do not play beep to caller ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys SPA-9000
Where can I get more info on this product? Thanks robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Thursday, December 29, 2005 1:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linksys SPA-9000 Kerry - we certainly have the ability to match any pricing you are receiving from Atacomm, we offer discounted VAR pricing but you need to have an account set up with an inside sales rep, and I can facilitate that. We also offer RMA support and firmware for the duration of manufacturer warranty on all the products we sell, whereas Atacomm only offers a 30 day warranty. We will be offering the SPA-9000, and are looking to contract with a few SP's, so we can upsell voicemail provisioning in conjunction with the units. We have the ability, but not the desire, to host the VM internally, as we are not a service provider, and don't want the potential headaches that come along with that. We may simply decide to act as a fulfillment and marketing agent for an SP, or group of SP's, and when we sell the SPA-9000, we are selling it on your behalf. We will likely offer pre-provisionment of the unit for customers that want that, and many customers don't want to deal with configuring the system even though it is fairly straightforward. We could probably set up some automated provisioning setup, so that you could remotely provision the SPA-9000 to the clients spec, and then we package and ship it. We can also do the provisionment in house, we provide outsourced SIP device provisionment and fulfillment for a wide variety of VOIP service providers, including DeltaThree, iConnectHere, Broadvoice, and hopefully soon, Vonage. We have a web based form where a client can outline how they want their autoattendant(s), extensions and other options configured. We have voice talent for prompts, or clients can provide their own. We offer a wide spectrum of value adds, including leasing and finance options, because as you know there is not a ton of margin in this hardware. If you have some time to chat later this week, I am anxious to see how SP's foresee pricing the voicemail service for the SPA-9000, on a per seat model, or a per pbx model. Let me know a convenient time to reach you. We are certainly looking for strong partnerships and we bring a lot to the table, with sales approaching $50MM and aggregating, on average, around 2200 new customers per month over the past year. Thanks for the email! Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Kerry Garrison wrote: Cory, Sherman at Linksys suggested I touch bases with you. We have an SPA-9000 here that we are testing out. We will be rolling out a voicemail service to go along with it as well. We have a small IT firm in southern California and are growing our IP PBX business quite nicely this year and expect 2006 to be very nice. We have been buying strictly from atacomm because of their prices but would rather have a good partnership with someone that can potentially help us out as well if you need help on the west coast. Just thought I would make the introduction and see if we can start talking about how we can work together. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys SPA-9000
In a few days it will be publicly announced. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Robert Augustyn Sent: Thursday, December 29, 2005 8:26 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Linksys SPA-9000 Where can I get more info on this product? Thanks robert -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cory Andrews Sent: Thursday, December 29, 2005 1:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Linksys SPA-9000 Kerry - we certainly have the ability to match any pricing you are receiving from Atacomm, we offer discounted VAR pricing but you need to have an account set up with an inside sales rep, and I can facilitate that. We also offer RMA support and firmware for the duration of manufacturer warranty on all the products we sell, whereas Atacomm only offers a 30 day warranty. We will be offering the SPA-9000, and are looking to contract with a few SP's, so we can upsell voicemail provisioning in conjunction with the units. We have the ability, but not the desire, to host the VM internally, as we are not a service provider, and don't want the potential headaches that come along with that. We may simply decide to act as a fulfillment and marketing agent for an SP, or group of SP's, and when we sell the SPA-9000, we are selling it on your behalf. We will likely offer pre-provisionment of the unit for customers that want that, and many customers don't want to deal with configuring the system even though it is fairly straightforward. We could probably set up some automated provisioning setup, so that you could remotely provision the SPA-9000 to the clients spec, and then we package and ship it. We can also do the provisionment in house, we provide outsourced SIP device provisionment and fulfillment for a wide variety of VOIP service providers, including DeltaThree, iConnectHere, Broadvoice, and hopefully soon, Vonage. We have a web based form where a client can outline how they want their autoattendant(s), extensions and other options configured. We have voice talent for prompts, or clients can provide their own. We offer a wide spectrum of value adds, including leasing and finance options, because as you know there is not a ton of margin in this hardware. If you have some time to chat later this week, I am anxious to see how SP's foresee pricing the voicemail service for the SPA-9000, on a per seat model, or a per pbx model. Let me know a convenient time to reach you. We are certainly looking for strong partnerships and we bring a lot to the table, with sales approaching $50MM and aggregating, on average, around 2200 new customers per month over the past year. Thanks for the email! Regards, Cory Andrews Senior Partner +++ VOIPSupply.com 454 Sonwil Drive Buffalo, NY 14225 +++ voice - 716.630.1555 X22 email - [EMAIL PROTECTED] fax - 716.630.1548 Kerry Garrison wrote: Cory, Sherman at Linksys suggested I touch bases with you. We have an SPA-9000 here that we are testing out. We will be rolling out a voicemail service to go along with it as well. We have a small IT firm in southern California and are growing our IP PBX business quite nicely this year and expect 2006 to be very nice. We have been buying strictly from atacomm because of their prices but would rather have a good partnership with someone that can potentially help us out as well if you need help on the west coast. Just thought I would make the introduction and see if we can start talking about how we can work together. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CoreDump
we have some problems with asterisk coredumping. We are running 1.0.9 on an Linux Debian Sarge with 2.4.31 Kernel. Inside is an wct4xxp (4 E1s). We terminate: SIP = Asterisk = DSS1 (gdb) bt #0 0x4052c831 in q921_transmit_iframe (pri=0x401e0938, buf=0xbe5ff444, len=9, cr=1) at q921.c:384 #1 0x40532224 in q931_xmit (pri=0x401e0938, h=0xbe5ff444, len=9, cr=1) at q931.c:1848 #2 0x40532401 in send_message (pri=0x401e0938, c=0x82b1b50, msgtype=77, ies=0x4053e800) at q931.c:1888 #3 0x40532d83 in q931_release (pri=0x401e0938, c=0x82b1b50, cause=16) at q931.c:2141 #4 0x40532b2d in pri_disconnect_timeout (data=0x82b1b50) at q931.c:2092 #5 0x4052e355 in __pri_schedule_run (pri=0x8185620, tv=0xbe5ff90c) at prisched.c:97 #6 0x4052e3c0 in pri_schedule_run (pri=0x8185620) at prisched.c:109 #7 0x404f9d8d in pri_dchannel (vpri=0x4050f000) at chan_zap.c:7415 #8 0x400200ba in pthread_start_thread () from /lib/libpthread.so.0 is this a known problem and we should switch an other version? best regards Markus ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] billing system
Yes. [EMAIL PROTECTED] wrote: Hello All, Have anybody test ISP BILLING SYSTEM ? http://ibs.sourceforge.net/index.html Regards Harry ___ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.2 + DMZ + NAT clients
Hey everyone I have my Asterisk server setup as the DMZ on my Linksys router. If I use the internal IP as the domain in Xlite clients will register and work, however, if I use the FQDN for my asterisk server the clients will not register. I have all the extensions set to NAT=yes and have modified sip.conf to include externip=insert FQDN here, externhost=insert FQDN here, and localnet=192.168.1.0/255.255.255.0 I see Xlite trying to register, but it never does. I've done some searching around on forums and this seems to be a fairly common setup that works for people, not sure why it won't work for me!Thanks,Blake ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does Page application do?
Paging is a quite common feature on modern PABX's and means that anyone can connect to any speakerphone to broadcast messages, in some cases even if the phone is in use. The typical usage would be the company secretary desperatly trying to get hold of someone and since that person don't answer the phone she makes a broadcast call to a department or to the entire company through the speaker-phones. I am not familiar with the Asterisk implementation! Jan Robert La Ferla wrote: Why would you use this? Can someone please elaborate on the below description? I'm missing the intent of it. localhost*CLI show application Page [Synopsis] Pages phones [Description] Page(Technology/ResourceTechnology2/Resource2[|options]) Places outbound calls to the given technology / resource and dumps them into a conference bridge as muted participants. The original caller is dumped into the conference as a speaker and the room is destroyed when the original caller leaves. Valid options are: d - full duplex audio q - quiet, do not play beep to caller ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk SIP PORTS
Hi I am running asterisk SIP on port 5060, in my sipura i changed the 5060 port to 6060. but it's still tring to register it to asterisk. how come this is possible, Regards Kani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does Page application do?
Want to page all the SNOM phones in the office? Create a second SIP
account set to auto answer.
OFFICE=SIP/501SIP/502SIP/503SIP/504SIP/505SIP/506SIP/507
[default]
; Paging - Office only
exten = 44,1,NoOp(Paging the office)
exten = 44,n,SIPAddHeader,Call-Info: sip:192.168.20.1/; anwser-after=0
exten = 44,n,Page(${OFFICE}|q)
very usefull if the client moves from a small office to a much larger
one and the owner wants to yell at everyone at once.
On 12/29/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Paging is a quite common feature on modern PABX's and means that anyone
can connect to any speakerphone to broadcast messages, in some cases
even if the phone is in use. The typical usage would be the company
secretary desperatly trying to get hold of someone and since that person
don't answer the phone she makes a broadcast call to a department or to
the entire company through the speaker-phones.
I am not familiar with the Asterisk implementation!
Jan
Robert La Ferla wrote:
Why would you use this? Can someone please elaborate on the below
description? I'm missing the intent of it.
localhost*CLI show application Page
[Synopsis]
Pages phones
[Description]
Page(Technology/ResourceTechnology2/Resource2[|options])
Places outbound calls to the given technology / resource and dumps
them into a conference bridge as muted participants. The original
caller is dumped into the conference as a speaker and the room is
destroyed when the original caller leaves. Valid options are:
d - full duplex audio
q - quiet, do not play beep to caller
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--
---
Andrew Latham - AKA: LATHAMA (lay-th-ham-eh)
[EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED]
If any of the above are down we have bigger problems than my email!
---
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RE: [Asterisk-Users] Asterisk 1.2 + DMZ + NAT clients
If the machines with X-Lite are on the local network, use the private ip, if they are outside the network, use the public ip. -Kerry From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Blake KroneSent: Thursday, December 29, 2005 9:49 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Asterisk 1.2 + DMZ + NAT clients Hey everyone I have my Asterisk server setup as the DMZ on my Linksys router. If I use the internal IP as the domain in Xlite clients will register and work, however, if I use the FQDN for my asterisk server the clients will not register. I have all the extensions set to NAT=yes and have modified sip.conf to include externip=insert FQDN here, externhost=insert FQDN here, and localnet=192.168.1.0/255.255.255.0 I see Xlite trying to register, but it never does. I've done some searching around on forums and this seems to be a fairly common setup that works for people, not sure why it won't work for me!Thanks,Blake ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] PRI Hangup cause
Hi all, I have a couple of LD PRI through Broadwing. I'm trying to verify that I get the correct cause codes during the hangup. Specifically, I want to know when a number is disconnected. All of the numbers I have tried give cause 16. I have gotten a number to give cause 31. Does someone have a list of disconnected numbers that I can go through to use for testing? These can't just be numbers from a VOIP provider or something like that who just play their own disconnected message and hangup. I need it to show the correct cause code for a disconnected number. I have tried: 724-287-9021, 734-655-1212, 850-697-6330. All of those give cause 16. Can someone with a LD PRI test those for me and tell me what cause code they get? Thanks, Kevin ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 + DMZ + NAT clients
Anyway around that? It's a PITA to have to change that all the time with my PDA laptop.On 12/29/05, Kerry Garrison [EMAIL PROTECTED] wrote: If the machines with X-Lite are on the local network, use the private ip, if they are outside the network, use the public ip. -Kerry From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Blake KroneSent: Thursday, December 29, 2005 9:49 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: [Asterisk-Users] Asterisk 1.2 + DMZ + NAT clients Hey everyone I have my Asterisk server setup as the DMZ on my Linksys router. If I use the internal IP as the domain in Xlite clients will register and work, however, if I use the FQDN for my asterisk server the clients will not register. I have all the extensions set to NAT=yes and have modified sip.conf to include externip=insert FQDN here, externhost=insert FQDN here, and localnet=192.168.1.0/255.255.255.0 I see Xlite trying to register, but it never does. I've done some searching around on forums and this seems to be a fairly common setup that works for people, not sure why it won't work for me!Thanks,Blake ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] CALLERIDNUM
www.voip-info.org/wiki-asterisk
or you could try the CLI show application Set, and show function CALLERID
On 12/28/05, Rehan Ahmed [EMAIL PROTECTED] wrote:
Hi
Can you send any example of this command like
Set(CALLERID(num)=value)
Thanks
Rehan
On 12/28/05, C F [EMAIL PROTECTED] wrote:
in 1.2 and on (or CVS HEAD) you have to use: Set(CALLERID(num)=value)
On 12/28/05, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
is it possible rewrite CALLERIDNUM in the ZAP channel? I use
[int-transfer]
exten = _00.,1,SetVar(CALLERIDNUM=${CALLNR})
exten = _00.,2,MYSQL(Connect connid localhost webcdr ser91623 cdr)
exten = _00.,3,MYSQL(Query resultid ${connid} select\
if((floor(u.credit/p.cost))1\,ceil((u.credit)/p.cost)*60\,0)\ as\
sekund\ from\ user\ u\,\ sip\ s\,\ pricelist\ p\ where\
u.iduser=s.iduser\ and\ s.idsip=\'${CALLERIDNUM}\'\ and\
p.acode=s.acode\ and\ u.currency=p.currency\ and\
right(left(\'${EXTEN}\'\,CHAR_LENGTH(
p.ccode)+2)\,CHAR_LENGTH(p.ccode))\
like\ concat(p.ccode\,\'%\')\ order\ by\ p.ccode\ desc\ limit\ 1)
exten = _00.,4,MYSQL(Fetch foundRow ${resultid} sekund) ; fetch
row
..
..
without success. At row 3 have var ${CALLERIDNUM} original value,
not value from ${CALLNR}.
--
[EMAIL PROTECTED]
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--
Rehan Ahmed AllahWala
http://www.SuperTec.com - Tommrow's Technology, Today.
http://www.didx.net - DID Number Exchange and Peering Service.
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Re: [Asterisk-Users] PRI: This number has been disconnected
I have tried both inband and outofband, doesn't seem to make a difference. I added the congension and playtones(congestion) to the dial plan after the dial, but the users just get a busy instead of Do-De-Dah The number of have reached is notin service fastbusy. PRI Debug below. -- Executing Dial(IAX2/sycam-16385, Zap/g2/8157872800) in new stack-- Making new call for cr 32816 -- Requested transfer capability: 0x00 - SPEECH Protocol Discriminator: Q.931 (8) len=46 Call Ref: len= 2 (reference 48/0x30) (Originator) Message type: SETUP (5) [04 03 80 90 a2] Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0) Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16) Ext: 1 User information layer 1: u-Law (34) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ] [1e 02 80 83]I Progress Indicator (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: User (0) Ext: 1 Progress Description: Calling equipment is non-ISDN. (3) ] [6c 0c 41 80 38 31 35 37 35 34 38 38 32 33] Calling Number (len=14) [ Ext: 0 TON: Subscriber Number (4) NPI: ISDN/Telephony Numbering Plan ( E.164/E.163) (1) Presentation: Presentation permitted, user number not screened (0) '8157548823' ] [70 0b a1 38 31 35 37 38 37 32 38 30 30] Called Number (len=13) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan ( E.164/E.163) (1) '8157872800' ] -- Called g2/8157872800 Protocol Discriminator: Q.931 (8) len=10 Call Ref: len= 2 (reference 48/0x30) (Terminator) Message type: CALL PROCEEDING (2) [18 03 a9 83 81] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 1 ]-- Processing IE 24 (cs0, Channel Identification) -- Zap/25-1 is proceeding passing it to IAX2/sycam-16385 Protocol Discriminator: Q.931 (8) len=9 Call Ref: len= 2 (reference 48/0x30) (Terminator) Message type: DISCONNECT (69) [08 02 82 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Public network serving the local user (2) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ]-- Processing IE 8 (cs0, Cause) -- Channel 0/1, span 2 got hangup requestNEW_HANGUP DEBUG: Calling q931_hangup, ourstate Disconnect Indication, peerstate Disconnect Request Protocol Discriminator: Q.931 (8) len=18 Call Ref: len= 2 (reference 48/0x30) (Originator) Message type: RELEASE (77) [08 02 81 81] Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) 0: 0 Location: Private network serving the local user (1) Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ] [7e 07 04 58 0b 2d 08 31 35] User-User Information (len= 9) [ 04 58 0b 2d 08 31 35 ] -- Hungup 'Zap/25-1' == No one is available to answer at this time (1:0/0/0) -- Executing PlayTones(IAX2/sycam-16385, congestion) in new stack -- Executing Congestion(IAX2/sycam-16385, ) in new stack == Spawn extension (pri, 7872800, 8) exited non-zero on 'IAX2/sycam-16385' -- Hungup 'IAX2/sycam-16385' Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 48/0x30) (Terminator) Message type: RELEASE COMPLETE (90)NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate NullNEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null On 12/29/05, Adam Goryachev [EMAIL PROTECTED] wrote: On Wed, 2005-12-28 at 14:00 -0300, Javier Ergas wrote: I believe this behavior has nothing to do with the [EMAIL PROTECTED] Scripts. I think the problem is in the PRI signalization. I can see the zap hangup messages when trying to call a disconnected number. . -- Executing Dial(SIP/9349-1787, ZAP/g0/2514990) in new stack -- Called g0/2514990 -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' == No one is available to answer at this time -- Executing Goto(SIP/9349-1787, s-NOANSWER|1) in new stack -- Goto (macro-dialout-trunk,s-NOANSWER,1) The telco says they are sending inband information with the status of the call, but Asterisk is hanging up the channel instead of connecting it to let hear the audio message. There is a post with a similar issue here: http://mailgate.supereva.com/comp/comp.dcom.isdn.capi/msg04138.html Is anyone experiencing the same behavior?Sounds like the difference between doing inband signalling or out ofband signalling. I think by default, a PRI uses out of band signalling, ie, it just sends a message saying this number if un reachable soasterisk just hangs up and plays the local congestion dialplan.What you need to do is use inband signalling, so that asterisk won't hangup, and instead will pass the audio from the telco through.See /etc/asterisk/zapata.conf:; PRI Out of band indications.; Enable this to report Busy and Congestion on a PRI using out-of-band; notification. Inband indication, as used by Asterisk doesn't seem to work;
Re: [Asterisk-Users] What does Page application do?
So I can set it up to call a bunch of extensions and broadcast a message to them without the user picking up? Can I do this with Aastra phones? This would be great for announcing incoming calls. You have a call from XXX . Press 1 to pickup Press 2 to send them to voicemail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Re: Go directly to new messages fromVoiceMailMain?
On 12/29/05, Alexander Lopez [EMAIL PROTECTED] wrote: I believe that there currently is no option for Auto-play You would have to edit the source code for that. That is correct. But, I think it's a good idea, so be looking for a bug on Mantis shortly to provide a new option for it. :) -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Problem getting D channel up on Sangoma A102
Hi all, I am installing an Asterisk box equipped with the Sangoma A102 card. The telco just tested the PRI interface and it is ll ok. I now connect my Asterisk box and I can't get the D-Channel up. If I enable intense pri debug I see messages like the following: --SNIP START-- [ 02 01 7f ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data -- Got SABME from network peer. Sending Unnumbered Acknowledgement [ 02 01 73 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter == Primary D-Channel on span 1 up pbx*CLI [ 02 01 7f ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data -- Got SABME from network peer. Sending Unnumbered Acknowledgement [ 02 01 73 ] Unnumbered frame: SAPI: 00 C/R: 1 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 0 11: 3 [ UA (unnumbered acknowledgement) ] 0 bytes of data -- Restarting T203 counter -- Restarting T203 counter == Primary D-Channel on span 1 up T203 counter expired, sending RR and scheduling T203 again Sending Receiver Ready (0) [ 00 01 01 01 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 000 P/F: 1 0 bytes of data -- Restarting T203 counter -- Retrying poll with f-bit Sending Receiver Ready (0) [ 00 01 01 01 ] Supervisory frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 Zero: 0 S: 0 01: 1 [ RR (receive ready) ] N(R): 000 P/F: 1 0 bytes of data -- Restarting T203 counter Stopping T_203 timer T_200 timer already going (3) Protocol Discriminator: Q.931 (8) len=13 Call Ref: len= 2 (reference 0/0x0) (Originator) Message type: RESTART (70) [18 03 a9 83 86] Channel ID (len= 5) [ Ext: 1 IntID: Implicit, PRI Spare: 0, Exclusive Dchan: 0 ChanSel: Reserved Ext: 1 Coding: 0 Number Specified Channel Type: 3 Ext: 1 Channel: 6 ] [79 01 80] Restart Indentifier (len= 3) [ Ext: 1 Spare: 0 Resetting Indicated Channel (0) ] -- T200 counter expired, What to do... -- Retransmitting 17 bytes [ 00 01 00 01 08 02 00 00 46 18 03 a9 83 86 79 01 80 ] Informational frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 N(S): 000 0: 0 N(R): 000 P: 1 13 bytes of data -- Rescheduling retransmission (2) -- T200 counter expired, What to do... -- Timeout occured, restarting PRI Sending Set Asynchronous Balanced Mode Extended [ 00 01 7f ] Unnumbered frame: SAPI: 00 C/R: 0 EA: 0 TEI: 000EA: 1 M3: 3 P/F: 1 M2: 3 11: 3 [ SABME (set asynchronous balanced mode extended) ] 0 bytes of data == Primary D-Channel on span 1 down --SNIP END-- Config is the following: zaptel.conf: span=1,1,2,esf,b8zs bchan=1-23 dchan=24 loadzone = us defaultzone=us zapata.conf [channels] language=fr context=from-pstn switchtype=national resetinterval=never signalling=pri_cpe faxdetect=incoming usecallerid=yes echocancel=yes echocancelwhenbridged=no echotraining=800 group=1 channel=1-23 Any hints appreciated Andre ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Gnet VP168S
Has anyone used a Gnet VP168S with Asterisk? I've been testing with softphones, and this will be my first attempt at using a hardware product to connect a standard POTS telephone. The limited specs I found online suggest it should work (SIP one FXS port and one PSTN Fall back port, but like I said, this is my first attempt at using hardward (and I've only been playing with Asterisk/softphones for 5-6 days). Any input would be appreciated. Tim Johnson - This message was sent using the fabrysociety.org Webmail. For more Information please visit ; http://www.fabrysociety.org ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] TDM2400 wierdness
The toneduration setting seems to have fixed it. Thanks for the tip! -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Vile Sent: Thursday, December 29, 2005 7:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM2400 wierdness Try adding a w in your dial statement. Asterisk will dial even if the line is not ready with a dialtone, adding a w will wait a bit and then dial the number. On 12/29/05, Kerry Garrison [EMAIL PROTECTED] wrote: Thanks, I will try that. -Kerry -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Roger Hill Sent: Thursday, December 29, 2005 1:09 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] TDM2400 wierdness Kerry: I hope this helps. I had EXACTLY the same symptom when I was trying to get an X100P clone to work yesterday. Bumping the toneduration parameter in zapata.conf to 200 milliseconds cured the problem. Roger Kerry Garrison wrote: Asterisk 1.2.1 Updated the TDM2400 driver over the weekend Incoming calls seem to work perfectly Outbound calls never connect. If you listen in on the call to a 7 digit local number, you hear the first 6 digits, then a small delay, then the last digit. Then there is a long pause before the line is picked up, then a very long pause before the telco fires back you call could not be completed at this time. Calling using an analog phone on that line works fine. Do I possibly have some DTMF issues or something like that? Any suggestions would be appreciated. This is my only installation with the TDM2400 so I am kind of at a loss. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Roger Hill 07739 707 180 Perseverance is the hard work you do after you get tired of doing the hard work you already did. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI: This number has been disconnected
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[Asterisk-Users] Realtime Multiple Asterisk boxes and rtcachefriends MWI
I am working an a multiple box asterisk solution. I need phones to be able to login to multiple asterisk servers. I need Phone A to be able to register to switch A and call Phone B that is registered to switch B. With rtcachfriends=no this can be done, However I then loss MWI and sip show peers plus if a Phone becomes unreachable the phone I get dead air until the dial timeout reached. With rtcachfriends=yes I get MWI Sip show peers, However I cannot call phones that register to a different switch. My current working solution is to have rtcachfriends=yes. Place the call via sip if dialstatus=chanunavaliable I then routethe call to the other switch via an IAX trunk. Everything works but I don't have a true load balance soltuion. Plus it really only works for 2 boxes. It get out of hand when I add more.. I have tried using AGI and dialing the full contact found in the SIP realtime table. It works if the phone is active, but if the phone is no active I get dead air until the dial timeout is reached. This will not work as I cannot have 12 sec of dead air. So is there a way know the status of a SIP UA? It is it in the SIP realtime data? I looked at regseconds but it does not seem to be it because I can have a UA that is unreachable and the regseconds are not expired. Could realtime be altered to add a status filed to the SIP realtime table? Or is there a asterisk configuration option that I missed? This is my first post so please forgive me if I posted this in the wrong list. Many thanks! Doug Gillespie___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] [EMAIL PROTECTED]
1GB gratis, Antivirus y Antispam Correo Yahoo!, el mejor correo web del mundo Abrí tu cuenta aquí___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does Page application do?
I think most all of the phones have a hack to get it working. Aastra analog ADSI phones even work as I read some where... On 12/29/05, Robert La Ferla [EMAIL PROTECTED] wrote: So I can set it up to call a bunch of extensions and broadcast a message to them without the user picking up? Can I do this with Aastra phones? This would be great for announcing incoming calls. You have a call from XXX . Press 1 to pickup Press 2 to send them to voicemail. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] PRI: This number has been disconnected
I have tried both inband and outofband too unsuccessfully. I think the priindication parameter says how Asterisk reports Busy and Congestion to the PSTN, not the other way around. In the Asterisk config sirrix.conf (http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sirrix.conf) there is a providetones parameter, witch I think handles the way that interface receives the signalization from the PSTN, but I think it wont work for zaptel/Zapata. Today I tried Asterisk 1.2 in another Telco and I experienced the same behavior. Im starting to think this is a bug in the Asterisk E1 signalization. De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] En nombre de Joe Pukepail Enviado el: Jueves, 29 de Diciembre de 2005 15:22 Para: Asterisk Users Mailing List - Non-Commercial Discussion Asunto: Re: [Asterisk-Users] PRI: This number has been disconnected I have tried both inband and outofband, doesn't seem to make a difference. I added the congension and playtones(congestion) to the dial plan after the dial, but the users just get a busy instead of Do-De-Dah The number of have reached is notin service fastbusy. PRI Debug below. On 12/29/05, Adam Goryachev [EMAIL PROTECTED] wrote: On Wed, 2005-12-28 at 14:00 -0300, Javier Ergas wrote: I believe this behavior has nothing to do with the [EMAIL PROTECTED] Scripts. I think the problem is in the PRI signalization. I can see the zap hangup messages when trying to call a disconnected number. . -- Executing Dial(SIP/9349-1787, ZAP/g0/2514990) in new stack -- Called g0/2514990 -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' == No one is available to answer at this time -- Executing Goto(SIP/9349-1787, s-NOANSWER|1) in new stack -- Goto (macro-dialout-trunk,s-NOANSWER,1) The telco says they are sending inband information with the status of the call, but Asterisk is hanging up the channel instead of connecting it to let hear the audio message. There is a post with a similar issue here: http://mailgate.supereva.com/comp/comp.dcom.isdn.capi/msg04138.html Is anyone experiencing the same behavior? Sounds like the difference between doing inband signalling or out of band signalling. I think by default, a PRI uses out of band signalling, ie, it just sends a message saying this number if un reachable so asterisk just hangs up and plays the local congestion dialplan. What you need to do is use inband signalling, so that asterisk won't hangup, and instead will pass the audio from the telco through. See /etc/asterisk/zapata.conf: ; PRI Out of band indications. ; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to work ; outofband:Signal Busy/Congestion out of band with RELEASE/DISCONNECT ; inband: Signal Busy/Congestion using in-band tones priindication = outofband Regards, Adam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Realtime Multiple Asterisk boxes andrtcachefriends MWI
The word from Kevin Fleming and Digium is that the use of realtime to support multiple Asterisk boxes sharing sip is not supported or even known to work at this point. -Original Message-From: Asterisk [mailto:[EMAIL PROTECTED]Sent: Thursday, December 29, 2005 12:14 PMTo: asterisk-users@lists.digium.comSubject: [Asterisk-Users] Realtime Multiple Asterisk boxes andrtcachefriends MWI I am working an a multiple box asterisk solution. I need phones to be able to login to multiple asterisk servers. I need Phone A to be able to register to switch A and call Phone B that is registered to switch B. With rtcachfriends=no this can be done, However I then loss MWI and sip show peers plus if a Phone becomes unreachable the phone I get dead air until the dial timeout reached. With rtcachfriends=yes I get MWI Sip show peers, However I cannot call phones that register to a different switch. My current working solution is to have rtcachfriends=yes. Place the call via sip if dialstatus=chanunavaliable I then routethe call to the other switch via an IAX trunk. Everything works but I don't have a true load balance soltuion. Plus it really only works for 2 boxes. It get out of hand when I add more.. I have tried using AGI and dialing the full contact found in the SIP realtime table. It works if the phone is active, but if the phone is no active I get dead air until the dial timeout is reached. This will not work as I cannot have 12 sec of dead air. So is there a way know the status of a SIP UA? It is it in the SIP realtime data? I looked at regseconds but it does not seem to be it because I can have a UA that is unreachable and the regseconds are not expired. Could realtime be altered to add a status filed to the SIP realtime table? Or is there a asterisk configuration option that I missed? This is my first post so please forgive me if I posted this in the wrong list. Many thanks! Doug Gillespie ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spandsp fax
Ok, Everything was fine up to the moment to run patch apps_makefile.patch Then I got Hunk 1 of 2, on the line 98 of the Makefile. This is the Makefile.rej output. As you can see, the line 98 includes some + signs that are in the apps_makefile.patch. [EMAIL PROTECTED] apps]# cat Makefile.rej ** 94,103 rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so app_curl.so: app_curl.o $(CC) $(SOLINK) -o $@ $ $(CURLLIBS) app_sql_postgres.o: app_sql_postgres.c $(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres. o app_sql_postgres.c app_sql_postgres.so: app_sql_postgres.o $(CC) $(SOLINK) -o $@ $ -L/usr/local/pgsql/lib -lpq --- 98,113 rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so app_curl.so: app_curl.o $(CC) $(SOLINK) -o $@ $ $(CURLLIBS) + app_rxfax.so : app_rxfax.o + $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff + + app_txfax.so : app_txfax.o + $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff + app_sql_postgres.o: app_sql_postgres.c $(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres. o app_sql_postgres.c app_sql_postgres.so: app_sql_postgres.o $(CC) $(SOLINK) -o $@ $ -L/usr/local/pgsql/lib -lpq However there is no request to take those lines of that file. Carlos Alperin [EMAIL PROTECTED] From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rehan Ahmed Sent: Wednesday, December 28, 2005 7:43 PM To: Dov Bigio; Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] spandsp fax Which version of Asterisk are you using ? 1.2 had problems in Make file for me 1.0.9 worked with a charm. You can email me with the error you have, maybe I can help you Rehan On 12/28/05, Dov Bigio [EMAIL PROTECTED] wrote: I am using Red Hat 9, but I don't think this changes the procedure - Original Message - From: Carlos Alperin [EMAIL PROTECTED] To: 'Dov Bigio' [EMAIL PROTECTED]; 'Asterisk Users Mailing List -Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 8:24 PM Subject: RE: [Asterisk-Users] spandsp fax Don, The previous question I believe was what linux are you using? By the way, I would like to know that too, just I was trying to make this work for weeks with no success. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]] On Behalf Of Dov Bigio Sent: Tuesday, December 27, 2005 10:54 AM To: Kristof Hardy; Asterisk Users Mailing List - Non-CommercialDiscussion Subject: Re: [Asterisk-Users] spandsp fax Hi BJ, Kristof, It worked! I am using the version at http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21c/asterisk-1. 2.x/. I think I had bad symlinks on /usr/local/lib and by reading the tutorial on AsteriskGuru I found that... (The previously installed version of spandsp has been 0.0.3, but now you have installed version 0.0.2. The problem is that the installation of version 0.0.3 creates a symlink, which is not replaced by installation of version 0.0.2. So the symlink points to the library of version 0.0.3, which actually does not exist.). I simply deleted all files related to spandsp from this directory and installed it again! Thank you Dov - Original Message - From: Kristof Hardy [EMAIL PROTECTED] To: Dov Bigio [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-CommercialDiscussion asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 12:59 PM Subject: Re: [Asterisk-Users] spandsp fax Dov Bigio wrote: I am using Asterisk 1.2.1 and followed instructions on http://www.asteriskguru.com/tutorials/spandsp.html to install faxing capability on my server. what platform are you running on? (wich distro?) Does the make of the app_txfax and app_rxfax work out well? ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rehan Ahmed AllahWala http://www.SuperTec.com - Tommrow's Technology, Today. http://www.didx.net - DID Number Exchange and Peering Service. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does Page application do?
Can this work with any ADSI phone? Can you send some links. The documentation is quite hard to find.. Thanks Jacques Andrew Latham wrote: I think most all of the phones have a hack to get it working. Aastra analog ADSI phones even work as I read some where... On 12/29/05, Robert La Ferla [EMAIL PROTECTED] wrote: So I can set it up to call a bunch of extensions and broadcast a message to them without the user picking up? Can I do this with Aastra phones? This would be great for announcing incoming calls. "You have a call from XXX . Press 1 to pickup Press 2 to send them to voicemail." ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! --- ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp fax
Make it manually, because there is somme diff from 1.0.9 edit Makefile and add : everything after + Pierre Carlos Alperin wrote: Ok, Everything was fine up to the moment to run patch apps_makefile.patch Then I got Hunk 1 of 2, on the line 98 of the Makefile. This is the Makefile.rej output. As you can see, the line 98 includes some + signs that are in the apps_makefile.patch. [EMAIL PROTECTED] apps]# cat Makefile.rej ** 94,103 rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so app_curl.so: app_curl.o $(CC) $(SOLINK) -o $@ $ $(CURLLIBS) app_sql_postgres.o: app_sql_postgres.c $(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres. o app_sql_postgres.c app_sql_postgres.so: app_sql_postgres.o $(CC) $(SOLINK) -o $@ $ -L/usr/local/pgsql/lib -lpq --- 98,113 rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so app_curl.so: app_curl.o $(CC) $(SOLINK) -o $@ $ $(CURLLIBS) + app_rxfax.so : app_rxfax.o + $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff + + app_txfax.so : app_txfax.o + $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff + app_sql_postgres.o: app_sql_postgres.c $(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres. o app_sql_postgres.c app_sql_postgres.so: app_sql_postgres.o $(CC) $(SOLINK) -o $@ $ -L/usr/local/pgsql/lib -lpq However there is no request to take those lines of that file. Carlos Alperin [EMAIL PROTECTED] *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Rehan Ahmed *Sent:* Wednesday, December 28, 2005 7:43 PM *To:* Dov Bigio; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] spandsp fax Which version of Asterisk are you using ? 1.2 had problems in Make file for me 1.0.9 worked with a charm. You can email me with the error you have, maybe I can help you Rehan On 12/28/05, *Dov Bigio* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am using Red Hat 9, but I don't think this changes the procedure - Original Message - From: Carlos Alperin [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: 'Dov Bigio' [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]; 'Asterisk Users Mailing List -Non-Commercial Discussion' asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 8:24 PM Subject: RE: [Asterisk-Users] spandsp fax Don, The previous question I believe was what linux are you using? By the way, I would like to know that too, just I was trying to make this work for weeks with no success. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Dov Bigio Sent: Tuesday, December 27, 2005 10:54 AM To: Kristof Hardy; Asterisk Users Mailing List - Non-CommercialDiscussion Subject: Re: [Asterisk-Users] spandsp fax Hi BJ, Kristof, It worked! I am using the version at http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21c/asterisk-1. 2.x/. I think I had bad symlinks on /usr/local/lib and by reading the tutorial on AsteriskGuru I found that... (The previously installed version of spandsp has been 0.0.3, but now you have installed version 0.0.2. The problem is that the installation of version 0.0.3 creates a symlink, which is not replaced by installation of version 0.0.2. So the symlink points to the library of version 0.0.3, which actually does not exist.). I simply deleted all files related to spandsp from this directory and installed it again! Thank you Dov - Original Message - From: Kristof Hardy [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: Dov Bigio [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]; Asterisk Users Mailing List - Non-CommercialDiscussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 12:59 PM Subject: Re: [Asterisk-Users] spandsp fax Dov Bigio wrote: I am using Asterisk 1.2.1 and followed instructions on http://www.asteriskguru.com/tutorials/spandsp.html to install faxing capability on my server. what platform are you running on? (wich distro?) Does the make of the app_txfax and app_rxfax work out well? ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:
[Asterisk-Users] Linksys SPA-942
Anybody know the status of the Linksys SPA-942? Is it out yet? -Bill ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spandsp fax
Do I need to compile first the app_rxfax.c app_txfax.c to get the .so files? If the answer is yes, how I do that command, just I'm not and expert on GCC. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pierre Burton Sent: Thursday, December 29, 2005 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] spandsp fax Make it manually, because there is somme diff from 1.0.9 edit Makefile and add : everything after + Pierre Carlos Alperin wrote: Ok, Everything was fine up to the moment to run patch apps_makefile.patch Then I got Hunk 1 of 2, on the line 98 of the Makefile. This is the Makefile.rej output. As you can see, the line 98 includes some + signs that are in the apps_makefile.patch. [EMAIL PROTECTED] apps]# cat Makefile.rej ** 94,103 rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so app_curl.so: app_curl.o $(CC) $(SOLINK) -o $@ $ $(CURLLIBS) app_sql_postgres.o: app_sql_postgres.c $(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres. o app_sql_postgres.c app_sql_postgres.so: app_sql_postgres.o $(CC) $(SOLINK) -o $@ $ -L/usr/local/pgsql/lib -lpq --- 98,113 rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so app_curl.so: app_curl.o $(CC) $(SOLINK) -o $@ $ $(CURLLIBS) + app_rxfax.so : app_rxfax.o + $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff + + app_txfax.so : app_txfax.o + $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff + app_sql_postgres.o: app_sql_postgres.c $(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres. o app_sql_postgres.c app_sql_postgres.so: app_sql_postgres.o $(CC) $(SOLINK) -o $@ $ -L/usr/local/pgsql/lib -lpq However there is no request to take those lines of that file. Carlos Alperin [EMAIL PROTECTED] *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Rehan Ahmed *Sent:* Wednesday, December 28, 2005 7:43 PM *To:* Dov Bigio; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] spandsp fax Which version of Asterisk are you using ? 1.2 had problems in Make file for me 1.0.9 worked with a charm. You can email me with the error you have, maybe I can help you Rehan On 12/28/05, *Dov Bigio* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am using Red Hat 9, but I don't think this changes the procedure - Original Message - From: Carlos Alperin [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: 'Dov Bigio' [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]; 'Asterisk Users Mailing List -Non-Commercial Discussion' asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 8:24 PM Subject: RE: [Asterisk-Users] spandsp fax Don, The previous question I believe was what linux are you using? By the way, I would like to know that too, just I was trying to make this work for weeks with no success. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Dov Bigio Sent: Tuesday, December 27, 2005 10:54 AM To: Kristof Hardy; Asterisk Users Mailing List - Non-CommercialDiscussion Subject: Re: [Asterisk-Users] spandsp fax Hi BJ, Kristof, It worked! I am using the version at http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21c/asterisk-1. 2.x/. I think I had bad symlinks on /usr/local/lib and by reading the tutorial on AsteriskGuru I found that... (The previously installed version of spandsp has been 0.0.3, but now you have installed version 0.0.2. The problem is that the installation of version 0.0.3 creates a symlink, which is not replaced by installation of version 0.0.2. So the symlink points to the library of version 0.0.3, which actually does not exist.). I simply deleted all files related to spandsp from this directory and installed it again! Thank you Dov - Original Message - From: Kristof Hardy [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: Dov Bigio [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]; Asterisk Users Mailing List - Non-CommercialDiscussion asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 12:59 PM Subject: Re: [Asterisk-Users] spandsp fax Dov Bigio wrote: I am using Asterisk 1.2.1 and followed instructions on http://www.asteriskguru.com/tutorials/spandsp.html to install faxing capability on my server. what platform are you running on? (wich distro?) Does the make of the app_txfax and app_rxfax work out well?
Re: [Asterisk-Users] Linksys SPA-942
Not out, nor expected in the near term. Cory Andrews Purchasing Manager ++ VOIPSupply.com A Division of b2 Technologies 454 Sonwil Drive Buffalo, NY 14225 direct - 716.250.3402 mobile - 716.907.4054 email - [EMAIL PROTECTED] AIM - b2Cory - Original Message - From: Hunt, Bill [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Thursday, December 29, 2005 2:56 PM Subject: [Asterisk-Users] Linksys SPA-942 Anybody know the status of the Linksys SPA-942? Is it out yet? -Bill ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk 1.2 + DMZ + NAT clients
On Friday 30 December 2005 07:19, Blake Krone wrote: Hey everyone I have my Asterisk server setup as the DMZ on my Linksys router. If I use the internal IP as the domain in Xlite clients will register and work, however, if I use the FQDN for my asterisk server the clients will not register. I have all the extensions set to NAT=yes and have modified sip.conf to include externip=insert FQDN here, externhost=insert FQDN here, and localnet=192.168.1.0/255.255.255.0 On 12/29/05, Kerry Garrison [EMAIL PROTECTED] wrote: If the machines with X-Lite are on the local network, use the private ip, if they are outside the network, use the public ip. Anyway around that? It's a PITA to have to change that all the time with my PDA laptop. You could set up an internal DNS server that points the FQDN to your private IP. hads -- The world's great men have not commonly been great scholars, nor its great scholars great men. -- Oliver Wendell Holmes ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Linksys SPA-942
Try visiting CES next week, it might be announced there. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hunt, Bill Sent: Thursday, December 29, 2005 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Linksys SPA-942 Anybody know the status of the Linksys SPA-942? Is it out yet? -Bill ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] spandsp fax
If you check the AsteriskGuru.com tutorial about this, he explains how to edit this files manually.. it is really simple! - Original Message - From: Carlos Alperin [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, December 29, 2005 5:59 PM Subject: RE: [Asterisk-Users] spandsp fax Do I need to compile first the app_rxfax.c app_txfax.c to get the .so files? If the answer is yes, how I do that command, just I'm not and expert on GCC. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pierre Burton Sent: Thursday, December 29, 2005 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] spandsp fax Make it manually, because there is somme diff from 1.0.9 edit Makefile and add : everything after + Pierre Carlos Alperin wrote: Ok, Everything was fine up to the moment to run patch apps_makefile.patch Then I got Hunk 1 of 2, on the line 98 of the Makefile. This is the Makefile.rej output. As you can see, the line 98 includes some + signs that are in the apps_makefile.patch. [EMAIL PROTECTED] apps]# cat Makefile.rej ** 94,103 rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so app_curl.so: app_curl.o $(CC) $(SOLINK) -o $@ $ $(CURLLIBS) app_sql_postgres.o: app_sql_postgres.c $(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres. o app_sql_postgres.c app_sql_postgres.so: app_sql_postgres.o $(CC) $(SOLINK) -o $@ $ -L/usr/local/pgsql/lib -lpq --- 98,113 rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so app_curl.so: app_curl.o $(CC) $(SOLINK) -o $@ $ $(CURLLIBS) + app_rxfax.so : app_rxfax.o + $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff + + app_txfax.so : app_txfax.o + $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff + app_sql_postgres.o: app_sql_postgres.c $(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres. o app_sql_postgres.c app_sql_postgres.so: app_sql_postgres.o $(CC) $(SOLINK) -o $@ $ -L/usr/local/pgsql/lib -lpq However there is no request to take those lines of that file. Carlos Alperin [EMAIL PROTECTED] *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Rehan Ahmed *Sent:* Wednesday, December 28, 2005 7:43 PM *To:* Dov Bigio; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] spandsp fax Which version of Asterisk are you using ? 1.2 had problems in Make file for me 1.0.9 worked with a charm. You can email me with the error you have, maybe I can help you Rehan On 12/28/05, *Dov Bigio* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am using Red Hat 9, but I don't think this changes the procedure - Original Message - From: Carlos Alperin [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: 'Dov Bigio' [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]; 'Asterisk Users Mailing List -Non-Commercial Discussion' asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 8:24 PM Subject: RE: [Asterisk-Users] spandsp fax Don, The previous question I believe was what linux are you using? By the way, I would like to know that too, just I was trying to make this work for weeks with no success. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Dov Bigio Sent: Tuesday, December 27, 2005 10:54 AM To: Kristof Hardy; Asterisk Users Mailing List - Non-CommercialDiscussion Subject: Re: [Asterisk-Users] spandsp fax Hi BJ, Kristof, It worked! I am using the version at http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21c/asterisk-1. 2.x/. I think I had bad symlinks on /usr/local/lib and by reading the tutorial on AsteriskGuru I found that... (The previously installed version of spandsp has been 0.0.3, but now you have installed version 0.0.2. The problem is that the installation of version 0.0.3 creates a symlink, which is not replaced by installation of version 0.0.2. So the symlink points to the library of version 0.0.3, which actually does not exist.). I simply deleted all files related to spandsp from this directory and installed it again! Thank you Dov - Original Message - From: Kristof Hardy [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: Dov Bigio [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]; Asterisk Users
[Asterisk-Users] Regular modems?
Hi all, I'm a freshfaced asterisk n00b, and I've got a dumb question. (tm) I'm experimenting with an asterisk at home install on a spare machine here. It has a PCI modem installed in it. Zapatel seems to have recognized this and configured trunk ZAP/g0. It does not, however, seem to work. I'm wondering if this is supposed to work, or if non-digium modems just won't work? I'd really like to play around with this for a bit before I have to justify even $150 to my boss, who hopefully won't know about this until it's a nearly-functional replacement for our current PBX, which I'm getting fed up with. =] So, if it will work with a regular modem, that'd be great. Thanks, Matt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] spandsp fax
Ok, That is the place where I download the procedure, but I didn't found anything about editing the Makefiles. Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Dov Bigio Sent: Thursday, December 29, 2005 3:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] spandsp fax If you check the AsteriskGuru.com tutorial about this, he explains how to edit this files manually.. it is really simple! - Original Message - From: Carlos Alperin [EMAIL PROTECTED] To: 'Asterisk Users Mailing List - Non-Commercial Discussion' asterisk-users@lists.digium.com Sent: Thursday, December 29, 2005 5:59 PM Subject: RE: [Asterisk-Users] spandsp fax Do I need to compile first the app_rxfax.c app_txfax.c to get the .so files? If the answer is yes, how I do that command, just I'm not and expert on GCC. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pierre Burton Sent: Thursday, December 29, 2005 2:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] spandsp fax Make it manually, because there is somme diff from 1.0.9 edit Makefile and add : everything after + Pierre Carlos Alperin wrote: Ok, Everything was fine up to the moment to run patch apps_makefile.patch Then I got Hunk 1 of 2, on the line 98 of the Makefile. This is the Makefile.rej output. As you can see, the line 98 includes some + signs that are in the apps_makefile.patch. [EMAIL PROTECTED] apps]# cat Makefile.rej ** 94,103 rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so app_curl.so: app_curl.o $(CC) $(SOLINK) -o $@ $ $(CURLLIBS) app_sql_postgres.o: app_sql_postgres.c $(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres. o app_sql_postgres.c app_sql_postgres.so: app_sql_postgres.o $(CC) $(SOLINK) -o $@ $ -L/usr/local/pgsql/lib -lpq --- 98,113 rm -f $(DESTDIR)$(MODULES_DIR)/app_qcall.so app_curl.so: app_curl.o $(CC) $(SOLINK) -o $@ $ $(CURLLIBS) + app_rxfax.so : app_rxfax.o + $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff + + app_txfax.so : app_txfax.o + $(CC) $(SOLINK) -o $@ $ -lspandsp -ltiff + app_sql_postgres.o: app_sql_postgres.c $(CC) -pipe -I/usr/local/pgsql/include $(CFLAGS) -c -o app_sql_postgres. o app_sql_postgres.c app_sql_postgres.so: app_sql_postgres.o $(CC) $(SOLINK) -o $@ $ -L/usr/local/pgsql/lib -lpq However there is no request to take those lines of that file. Carlos Alperin [EMAIL PROTECTED] *From:* [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] *On Behalf Of *Rehan Ahmed *Sent:* Wednesday, December 28, 2005 7:43 PM *To:* Dov Bigio; Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [Asterisk-Users] spandsp fax Which version of Asterisk are you using ? 1.2 had problems in Make file for me 1.0.9 worked with a charm. You can email me with the error you have, maybe I can help you Rehan On 12/28/05, *Dov Bigio* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: I am using Red Hat 9, but I don't think this changes the procedure - Original Message - From: Carlos Alperin [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] To: 'Dov Bigio' [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]; 'Asterisk Users Mailing List -Non-Commercial Discussion' asterisk-users@lists.digium.com mailto:asterisk-users@lists.digium.com Sent: Tuesday, December 27, 2005 8:24 PM Subject: RE: [Asterisk-Users] spandsp fax Don, The previous question I believe was what linux are you using? By the way, I would like to know that too, just I was trying to make this work for weeks with no success. Thanks, Carlos Alperin -Original Message- From: [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] mailto:[EMAIL PROTECTED]] On Behalf Of Dov Bigio Sent: Tuesday, December 27, 2005 10:54 AM To: Kristof Hardy; Asterisk Users Mailing List - Non-CommercialDiscussion Subject: Re: [Asterisk-Users] spandsp fax Hi BJ, Kristof, It worked! I am using the version at http://www.soft-switch.org/downloads/spandsp/spandsp-0.0.2pre21c/asterisk-1. 2.x/. I think I had bad symlinks on /usr/local/lib and by reading the tutorial on AsteriskGuru I found that... (The previously installed version of spandsp has been 0.0.3, but now you have installed version 0.0.2. The problem is that the installation of version 0.0.3 creates a symlink, which is not replaced by installation of version 0.0.2. So the symlink
[Asterisk-Users] voicemail storage over odbc and postgres
Has anyone gotten voicemail storage over odbc working with postgres? I have been trying to get this working and keep hitting snags. Elazar ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Getting Yoda unit to register all four ports
I have a sample of the Yoda VG400 and I am having a devil of a time trying to get all four channels to register to Asterisk. I have an Asterisk 1.2.1 server. I have tried adding one at a time and rebooting it, but it stops after the first. http://www.yoda.com.tw/model.php?type=Enterprise_VoIPpname=VG400 Anyone had success with this? -- Chris Mason NetConcepts (264) 497-5670 Fax: (264) 497-8463 Int: (305) 704-7249 Fax: (815)301-9759 Cell: 264-235-5670 Yahoo IM: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] zaptel TDM21B 4-5 second pause
Try to append # or * to numberDial(ZAP/g0/0199255#) or Dial(ZAP/g0/0199255*)Cheers,Giovanni Miano 2005/12/29, Eck [EMAIL PROTECTED]:Hi, Sorry if this is a little off topic as its really more zaptel related, but hopefully someone will have come across this.,I am noticing a 4-5 second pause when my Digium TDM21B is dialing, just before dialing the last digit. This is causing me problems here in the UK as some telco (no prizes for guessing which one) seems to have reduced thier tolerence on DMTF pauses on some switches, so the switch is timing out after ten digits and not getting the eleventh because of the pause. The installation is [EMAIL PROTECTED] v2.2.an example prefix this is happining on is: +44(0)199255.I have worked around the problem by reducing DTMF_PAUSE in digits.h and recompliing zaptel, but this seems kludgey, does anyone know of a better solution? Many thanks for any help.-Alex.___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] sip debug file.txt
On Thu, Dec 29, 2005 at 12:51:47PM +0100, Olle E Johansson wrote: I usually do asterisk -rvn | tee /tmp/sipdebug.txt Then turn on sip debug on the cli. This captures everything. You need to make sure that the debug output is sent to the console in logger.conf script(1) would have given you something rather equivalent. However you still get bad escape sequences to filter out. Getting that from the logger is probably better. -- Tzafrir Cohen | [EMAIL PROTECTED] | VIM is http://tzafrir.org.il | | a Mutt's [EMAIL PROTECTED] | | best ICQ# 16849755 | | friend ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Server Hangs
whats kernel version ? check in dmesg for system messagesCheers,Giovanni Miano2005/12/29, Dushyanth Harinath [EMAIL PROTECTED] :Hey guys,Asterisk Server Specs :Asterisk version :CLI show version Asterisk SVN-trunk-r7230 built by [EMAIL PROTECTED] on a i686 running Linuxon 2005-12-25 16:14:47 UTCSystem details :Centos 4.2 (Final)Linux ip-pbx 2.6.9-22.ELsmp #1 SMPIntel Dual Xeon 3.06Ghz Intel SE7501CW2 MotherboardDigium cards : T110P (E1) , TDM22B, TDM31B, TDM24012BI added TDM24012B yes'day but haven't configured or used it yet. Itsjust connected to the system. The same problem used to occur before adding TDM24012B to the mix.This setup hangs up i,e total freeze cant ssh, cant login even from thesystem console and nothing in system logs or asterisk logs point me toany obvious problem. There is no coredump in /tmp too. Asterisk also freezes up the server when i issue a stop now command inthe CLI sometimes.The only call traffic at this moment are SIP to SIP internal calls, SIPto ZAP external calls and ZAP to SIP incoming calls. In all there must be a total of 10 simultaneous calls.Im using queues, rxfax, txfax, voicemail, meetme (still testing).This happens three or four times in a day.I cant see any IRQ misses in zttool and zttest output is below. Opened pseudo zap interface, measuring accuracy...99.987793% 99.987793% 99.987793% 99.987793% 100.00% 100.00%99.987793%99.987793% 100.00% 100.00% 100.00% 99.987793% 100.00%99.987793% 100.00%Best: 100.00 -- Worst: 99.987793 -- Average: 99.992300Found the below messages in dmesg but seems informational rather than aerror.Dec 27 22:04:24 asterisk kernel: zaptel Disabled echo canceller because of tone (tx) on channel 32Dec 29 21:02:12 asterisk kernel: zaptel Disabled echo canceller becauseof tone (rx) on channel 35I dont know what the problem could be. I followed the doc at http://www.voip-info.org/wiki-Asterisk+debugging and started asteriskusing safe_asterisk and applied the logger related changes.Wat else i can do to debug this issue ?Dushyanth___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Easiest way to use HFC-S?
Use Bristuff2005/12/29, Pisac [EMAIL PROTECTED]: What is the easiest way to install and use HFC-S card on Asterisk?As less kernel compiling driver installations as possible.Is it mISDN, or chan_capi, or vISDN, or zaphfc, or?___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Giovanni Miano ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] transfers using # in asterisk
Greetings fellow list members, I am using ABE and I am attempting to impliment transfers using "#". I am using both "T" and "t" as options in my Dial() command. I am attempting to hit "#" then enter another extension from my dialplan. I have tried this on both ends of the conversation and also tried hitting "#" again after entering the extension and still no luck. One end of the conversation is a SNOM 320, the other is an outside line. The transfer does not happen, I was wondering if anyone had any suggestions for me, perhaps something easily missed. I've looked at the wiki and I do have canreinvite set to no. Any help or ideas are much appreciated. Thank you, Frank Webb Inter Media Marketing Solutions ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Regular modems?
Use one of the clone X100 cards, available on eBay for 10 bucks or so Not all PCI Voice modems work Only ones with a certain chipset John Novack Matt Murphy wrote: Hi all, I'm a freshfaced asterisk n00b, and I've got a dumb question. (tm) I'm experimenting with an asterisk at home install on a spare machine here. It has a PCI modem installed in it. Zapatel seems to have recognized this and configured trunk ZAP/g0. It does not, however, seem to work. I'm wondering if this is supposed to work, or if non-digium modems just won't work? I'd really like to play around with this for a bit before I have to justify even $150 to my boss, who hopefully won't know about this until it's a nearly-functional replacement for our current PBX, which I'm getting fed up with. =] So, if it will work with a regular modem, that'd be great. Thanks, Matt ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a Gateway
Thanks James,
That should help to start my project Thanks a million...
I will keep on updating..
And thanks to all for the inputs
Thanks,
Neal
On Dec 29, 2005, at 6:39 AM, James Sizemore wrote:
Nitesh Divecha wrote:
Are there any examples of dial plans? Like how to make the default
context?
I just need a kick start on the config part, as I am really
struggling
on routing the calls.
Here is a very very simple example using a PRI. You will need more
error routing in a real dial plan:
extensions.conf:
[general]
static=yes
writeprotect=no
country=us
[local]
include = default
[globals]
TRUNK=Zap/g1
LDTRUNK=Zap/g2
[trunk]
;Long distance pstn
exten = _1NXXNXX,1,Dial(${LDTRUNK}/${EXTEN})
exten = _1NXXNXX,2,Hangup
;pstn
exten = _X.,1,Dial(${TRUNK}/${EXTEN})
exten = _X.,2,Hangup
[default-out]
;This is where you sent trusted calls from sip.conf out to pstn
include = trunk
[default]
;you send incoming pstn calls here as well as untrusted voip calls.
;here you would route call to local numbers you own via enum or
static.
exten = 6153247060,1,Wait(2) ; you need to wait
; long enough to get
; CNAM off line
;send incoming call to your register server.
exten = 55,2,Dial(SIP/[EMAIL PROTECTED])
sip.conf:
[general]
bindport = 5060
bindaddr = 0.0.0.0
context = default ; non trusted call from sip side go here
srvlookup = yes
dtmfmode=info
disallow=all
allow=ulaw
allow=alaw
allow=g729
[trusted]
type=friend
context=default-out ; trusted call can go out pstn
host=192.168.0.1
canreinvite=no
zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
span=3,1,0,esf,b8zs
bchan=49-71
dchan=72
span=4,1,0,esf,b8zs
bchan=73-95
dchan=96
loadzone = us
defaultzone=us
zapata.conf:
[channels]
context=default;pstn incoming call go here
switchtype=national
signalling=pri_cpe
toneduration=500
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=-1.0
txgain=-1.0
callerid=asreceived
;
group=1
channel=1-23
channel=73-95
;
group=2
channel=25-47
channel=49-71
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Nitesh Divecha
VoIP/Network Engineer
Viper Networks
10373 Roselle St. Ste:170
San Diego, CA. 92121
Phone: 858-452-8737
Fax: 858-452-8638
Cell: 1-909-964-5181
vPhone: 544-416-0067
Email: [EMAIL PROTECTED]
Web: www.vipernetworks.com
Your Internet Phone Company
A publicly traded Company, OTC: VPER
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[Asterisk-Users] Need Asterisk person in SoCal
I have a very interesting project to put together in the southern California area and am looking for anyone locally (preferably) that would be interested in being involved in it. I cannot go into too many details but it does require some good Asterisk configuration skills. Please email me off-list to discuss. Kerry Garrison Director of Technical Services Tech Data Pros - Orange County's Mobile IT Service Provider (949) 502-7819 x200 - [EMAIL PROTECTED] http://www.techdatapros.com ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a Gateway
The line that reads:
exten = 6153247060,1,Wait(2)
should have been:
exten = 55,1,Wait(2)
Nitesh Divecha wrote:
Thanks James,
That should help to start my project Thanks a million...
I will keep on updating..
And thanks to all for the inputs
Thanks,
Neal
On Dec 29, 2005, at 6:39 AM, James Sizemore wrote:
Nitesh Divecha wrote:
Are there any examples of dial plans? Like how to make the default
context?
I just need a kick start on the config part, as I am really
struggling
on routing the calls.
Here is a very very simple example using a PRI. You will need more
error routing in a real dial plan:
extensions.conf:
[general]
static=yes
writeprotect=no
country=us
[local]
include = default
[globals]
TRUNK=Zap/g1
LDTRUNK=Zap/g2
[trunk]
;Long distance pstn
exten = _1NXXNXX,1,Dial(${LDTRUNK}/${EXTEN})
exten = _1NXXNXX,2,Hangup
;pstn
exten = _X.,1,Dial(${TRUNK}/${EXTEN})
exten = _X.,2,Hangup
[default-out]
;This is where you sent trusted calls from sip.conf out to pstn
include = trunk
[default]
;you send incoming pstn calls here as well as untrusted voip calls.
;here you would route call to local numbers you own via enum or static.
exten = 6153247060,1,Wait(2) ; you need to wait
; long enough to get
; CNAM off line
;send incoming call to your register server.
exten = 55,2,Dial(SIP/[EMAIL PROTECTED])
sip.conf:
[general]
bindport = 5060
bindaddr = 0.0.0.0
context = default ; non trusted call from sip side go here
srvlookup = yes
dtmfmode=info
disallow=all
allow=ulaw
allow=alaw
allow=g729
[trusted]
type=friend
context=default-out ; trusted call can go out pstn
host=192.168.0.1
canreinvite=no
zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
span=3,1,0,esf,b8zs
bchan=49-71
dchan=72
span=4,1,0,esf,b8zs
bchan=73-95
dchan=96
loadzone = us
defaultzone=us
zapata.conf:
[channels]
context=default;pstn incoming call go here
switchtype=national
signalling=pri_cpe
toneduration=500
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=-1.0
txgain=-1.0
callerid=asreceived
;
group=1
channel=1-23
channel=73-95
;
group=2
channel=25-47
channel=49-71
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Nitesh Divecha
VoIP/Network Engineer
Viper Networks
10373 Roselle St. Ste:170
San Diego, CA. 92121
Phone: 858-452-8737
Fax: 858-452-8638
Cell: 1-909-964-5181
vPhone: 544-416-0067
Email: [EMAIL PROTECTED]
Web: www.vipernetworks.com
Your Internet Phone Company
A publicly traded Company, OTC: VPER
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RE: [Asterisk-Users] Regular modems?
Hmm, did a search, didn't come up with anything under X100 or clone X100, I assume you're talking about a few specific models, any ideas which? Thanks, Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Thursday, December 29, 2005 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Regular modems? Use one of the clone X100 cards, available on eBay for 10 bucks or so Not all PCI Voice modems work Only ones with a certain chipset John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Allison on Free 411
I heard on the radio about 1-800-FREE411andtried it out, Iwas very suprised to hear allisons' voicefor the digits. Not sure if theyare using asterisk for the backend on this or not. Try it out its Free! http://www.snopes.com/inboxer/nothing/free411.asp (not afflicated with it in any way). ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] PRI: This number has been disconnected
I am using T1 Signaling and seeing the same problems (I think), so I don't think its just E1. On 12/29/05, Javier Ergas [EMAIL PROTECTED] wrote: I have tried both inband and outofband too unsuccessfully. I think the priindication parameter says how Asterisk reports Busy and Congestion to the PSTN, not the other way around. In the Asterisk config sirrix.conf ( http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sirrix.conf) there is a providetones parameter, witch I think handles the way that interface receives the signalization from the PSTN, but I think it won't work for zaptel/Zapata. Today I tried Asterisk 1.2 in another Telco and I experienced the same behavior. I'm starting to think this is a bug in the Asterisk E1 signalization. De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] En nombre de Joe PukepailEnviado el: Jueves, 29 de Diciembre de 2005 15:22 Para: Asterisk Users Mailing List - Non-Commercial DiscussionAsunto: Re: [Asterisk-Users] PRI: This number has been disconnected I have tried both inband and outofband, doesn't seem to make a difference. I added the congension and playtones(congestion) to the dial plan after the dial, but the users just get a busy instead of Do-De-Dah The number of have reached is notin service fastbusy. PRI Debug below. … … On 12/29/05, Adam Goryachev [EMAIL PROTECTED] wrote: On Wed, 2005-12-28 at 14:00 -0300, Javier Ergas wrote: I believe this behavior has nothing to do with the [EMAIL PROTECTED] Scripts. I think the problem is in the PRI signalization. I can see the zap hangup messages when trying to call a disconnected number. . -- Executing Dial(SIP/9349-1787, ZAP/g0/2514990) in new stack -- Called g0/2514990 -- Channel 0/2, span 1 got hangup -- Hungup 'Zap/2-1' == No one is available to answer at this time -- Executing Goto(SIP/9349-1787, s-NOANSWER|1) in new stack -- Goto (macro-dialout-trunk,s-NOANSWER,1) The telco says they are sending inband information with the status of the call, but Asterisk is hanging up the channel instead of connecting it to let hear the audio message. There is a post with a similar issue here: http://mailgate.supereva.com/comp/comp.dcom.isdn.capi/msg04138.html Is anyone experiencing the same behavior?Sounds like the difference between doing inband signalling or out of band signalling. I think by default, a PRI uses out of band signalling, ie, it just sends a message saying this number if un reachable soasterisk just hangs up and plays the local congestion dialplan. What you need to do is use inband signalling, so that asterisk won't hangup, and instead will pass the audio from the telco through.See /etc/asterisk/zapata.conf:; PRI Out of band indications.; Enable this to report Busy and Congestion on a PRI using out-of-band ; notification. Inband indication, as used by Asterisk doesn't seem to work; outofband:Signal Busy/Congestion out of band withRELEASE/DISCONNECT; inband: Signal Busy/Congestion using in-band tones priindication = outofbandRegards,Adam___ --Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk as a Gateway
Just a quick one - did you do 'make samples' as part of installing asterisk?
That would have given you something to work with, at least.
(all of the files are in the configs folder in the Asterisk src folder if
you want to peruse them)
PaulH
- Original Message -
From: Nitesh Divecha [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, December 30, 2005 9:02 AM
Subject: Re: [Asterisk-Users] Asterisk as a Gateway
Thanks James,
That should help to start my project Thanks a million...
I will keep on updating..
And thanks to all for the inputs
Thanks,
Neal
On Dec 29, 2005, at 6:39 AM, James Sizemore wrote:
Nitesh Divecha wrote:
Are there any examples of dial plans? Like how to make the default
context?
I just need a kick start on the config part, as I am really
struggling
on routing the calls.
Here is a very very simple example using a PRI. You will need more
error routing in a real dial plan:
extensions.conf:
[general]
static=yes
writeprotect=no
country=us
[local]
include = default
[globals]
TRUNK=Zap/g1
LDTRUNK=Zap/g2
[trunk]
;Long distance pstn
exten = _1NXXNXX,1,Dial(${LDTRUNK}/${EXTEN})
exten = _1NXXNXX,2,Hangup
;pstn
exten = _X.,1,Dial(${TRUNK}/${EXTEN})
exten = _X.,2,Hangup
[default-out]
;This is where you sent trusted calls from sip.conf out to pstn
include = trunk
[default]
;you send incoming pstn calls here as well as untrusted voip calls.
;here you would route call to local numbers you own via enum or
static.
exten = 6153247060,1,Wait(2) ; you need to wait
; long enough to get
; CNAM off line
;send incoming call to your register server.
exten = 55,2,Dial(SIP/[EMAIL PROTECTED])
sip.conf:
[general]
bindport = 5060
bindaddr = 0.0.0.0
context = default ; non trusted call from sip side go here
srvlookup = yes
dtmfmode=info
disallow=all
allow=ulaw
allow=alaw
allow=g729
[trusted]
type=friend
context=default-out ; trusted call can go out pstn
host=192.168.0.1
canreinvite=no
zaptel.conf:
span=1,1,0,esf,b8zs
bchan=1-23
dchan=24
span=2,1,0,esf,b8zs
bchan=25-47
dchan=48
span=3,1,0,esf,b8zs
bchan=49-71
dchan=72
span=4,1,0,esf,b8zs
bchan=73-95
dchan=96
loadzone = us
defaultzone=us
zapata.conf:
[channels]
context=default;pstn incoming call go here
switchtype=national
signalling=pri_cpe
toneduration=500
usecallerid=yes
hidecallerid=no
callwaitingcallerid=yes
echocancel=yes
echocancelwhenbridged=yes
echotraining=800
rxgain=-1.0
txgain=-1.0
callerid=asreceived
;
group=1
channel=1-23
channel=73-95
;
group=2
channel=25-47
channel=49-71
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Nitesh Divecha
VoIP/Network Engineer
Viper Networks
10373 Roselle St. Ste:170
San Diego, CA. 92121
Phone: 858-452-8737
Fax: 858-452-8638
Cell: 1-909-964-5181
vPhone: 544-416-0067
Email: [EMAIL PROTECTED]
Web: www.vipernetworks.com
Your Internet Phone Company
A publicly traded Company, OTC: VPER
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Re: [Asterisk-Users] transfers using # in asterisk
Any idea what version of Asterisk ABE is based on? PaulH - Original Message - From: Franklin Webb To: asterisk-users@lists.digium.com Sent: Friday, December 30, 2005 8:43 AM Subject: [Asterisk-Users] transfers using # in asterisk Greetings fellow list members, I am using ABE and I am attempting to impliment transfers using "#". I am using both "T" and "t" as options in my Dial() command. I am attempting to hit "#" then enter another extension from my dialplan. I have tried this on both ends of the conversation and also tried hitting "#" again after entering the extension and still no luck. One end of the conversation is a SNOM 320, the other is an outside line. The transfer does not happen, I was wondering if anyone had any suggestions for me, perhaps something easily missed. I've looked at the wiki and I do have canreinvite set to no. Any help or ideas are much appreciated. Thank you, Frank Webb Inter Media Marketing Solutions ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] What does Page application do?
I think most all of the phones have a hack to get it working. Aastra analog ADSI phones even work as I read some where... Not the phones, but Asterisk needs to have a 'hack' to get this working. So, there must somethere i * code be a list of phones that has been implemented. PABX phones usually solved this because the PABX and phone vendor where the same. An analogue phone can be supported if it allows a broadcast to the speaker, which require that it has some intelligence to know that it needs to open the voice stream to the speaker even if the called has not picked up the phone. (The PABX can obviously intercept the voice stream and broadcast a message, but this is not really paging) H.323, SIP and many others have the same problem but the 'hack' is in these cases are a separate RTP stream to the speaker, without any call control, if this is allowerd (a majority of VoIP phones will allow 3 independant rtp streams) as ca 70% of the marked is Broadcom chips these days. I would expect that any phone which has this capability and that are commonly used would be supported in Asterisk? Jan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Regular modems?
Go to ebay, search on x100P, there are always several for sale. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matt Murphy Sent: Thursday, December 29, 2005 2:18 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Regular modems? Hmm, did a search, didn't come up with anything under X100 or clone X100, I assume you're talking about a few specific models, any ideas which? Thanks, Matt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Novack Sent: Thursday, December 29, 2005 4:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Regular modems? Use one of the clone X100 cards, available on eBay for 10 bucks or so Not all PCI Voice modems work Only ones with a certain chipset John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
