[asterisk-users] Providers that offer contract

Hello,I'm interested in contract some providers VoIP that support IAX2 and that offers the possibility of contract for outbound(termination) calls only. I'm not interested in DIDS, only termination calls.Thanks in advance ___ --Bandwidth and Colocation

Re: [asterisk-users] H323

On Sunday 27 August 2006 10:40, Mohammad Salaque wrote: any one try that with g723 codec? We use G.723.1, and it works well. My only problem is the bridging time (after pickup) takes at least 5 seconds. But this happenned even before Asterisk was in the picture, so I'm guessing it's the

[asterisk-users] Re: GSM gateway and FXO ATA

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 2) If the phone is answered on the first ring the call goes off to la la land. Explaining to users (or myself) that you need to wait for the second audible ring on the handset's before answering isn't acceptable. Hi Marty! Can you

Re: [asterisk-users] Selecting outbound trunk

On Tue, Aug 29, 2006 at 02:18:32PM +1000, Devraj Mukherjee wrote: The simplest way I can think of solving this is using prefixes, so someone appends a 0 or 1 and the dialplan puts the call through the selected trunk, where 0 being voip and 1 being PSTN. Whats wrong with something like this :

[asterisk-users] Re: Nokia E60/61/70 and SIP

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 1. you need qualify set as the wifi radio on the phone sucks big oranges What is qualify set? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail:

[asterisk-users] SIP Error message

Does anyone know what causes the following error message means. Aug 29 10:11:08 WARNING[30913]: chan_sip.c:2561 sip_write: Asked to transmit frame type 256, while native formats is 8 (read/write = 256/256) I've not yet tracked down what is causing this, but I get a lot of them at the same time.

Re: [asterisk-users] misdn-init.conf card parameter for a monoBRI

Patrick, thank you for your help, now I installed pciutils which is what I couldn't find so misdn-init works Unfortunately aftere a misdn-init start I get invalide module format errors. I search on internet hoping to solve it. Thanks again! Giorgio Incantalupo Patrick wrote: On Mon,

Re: [asterisk-users] Re: Nokia E60/61/70 and SIP

qualify=yes Put in in the sip.conf file in the configuration section for the specific phones. On Tue, 2006-08-29 at 09:50, Tomislav Parčina wrote: In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... 1. you need qualify set as the wifi radio on the phone sucks big oranges What is qualify

Re: [asterisk-users] misdn-init.conf card parameter for a monoBRI

what about a subscription on the Misdn-asterisk@lists.beronet.com mailing list! http://lists.beronet.com/cgi-bin/mailman/listinfo/misdn-asterisk Regards Kai ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list

[asterisk-users] Re: DNS

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Have you tried setting timeout, attempts and rotate in resolv.conf? Can you please tell me more about this? How to do it and what would I achieve with that? -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.:

[asterisk-users] sip giving problems, please help.

Can anyone explain this:- I am getting this on virtually every call. -- Executing Dial(SIP/15552830438-990b, SIP/[EMAIL PROTECTED]|) in new stack -- Called [EMAIL PROTECTED] -- SIP/192.168.1.1-859d is making progress passing it to SIP/15552830438-990b -- SIP/192.168.1.1-859d is

[asterisk-users] does misdn-mqueue work if compiled with gcc 4?

Hi, does misdn-mqueue work if compiled with gcc 4? I get some errors when trying to load misdn drivers: FATAL: Error inserting mISDN_core (/lib/modules/2.6.15-1-486/extra/mISDN_core.ko): Invalid module format WARNING: Error inserting mISDN_core (/lib/modules/2.6.15-1-486/extra/mISDN_core.ko):

[asterisk-users] transform bridged call into a conference

Hi! I wonder if it is possible to transform a bridged call into a conference. E.g. phone 1 calls phone 2 (normal bridged call with Dial()). Further phone 3 wants to join? Is this possible? Can you please refer me the proper applications? thanks klaus

[asterisk-users] Asterisk 1.2.4 I hear other party's voice only when I speack need help

I have this problem with Asterisk 1.2.4 I hear other party's voice only when I speack or i make some noise. Otherwise i hear nothing. Futhermore every time i receive a call , this message is displayed : -- Started music on hold, class 'my_class', on SIP/ some random public ip address

[asterisk-users] Asterisk - Comfort Noise

Hi, Does anybody know if asterisk 1.4 will support comfort noise? Or if there is a patch for it now? If it will be in 1.4 any idea of release date? Thanks, Dean Bath. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing

[asterisk-users] Re: H264

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... Actually you need to use the SVN version of Asterisk to support H264 video. It should be part of the planned 1.4 release. When can I expect 1.4 release? Will it be this year? First quarter of 2008? -- Tomislav Parčina Lama Computers

[asterisk-users] Mix Monitor call quality

Hi trying to record calls using mixmonitor, but I'm having problems with call quality the call seems OK but then it drops frames with silence ( for less than 0.5 seconds) then call continues All I'm doing is bridging two zap channels and recording no transcoding or changes to the channels

Re: [asterisk-users] does misdn-mqueue work if compiled with gcc 4?

gcc -v : gcc version 4.1.0 no problems using latest stuff from beronet/downloads/misdn_queue.stuff suse standard kernel regards KAI ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] does misdn-mqueue work if compiled with gcc 4?

Hi Kai, thanks, maybe I used the wrong kernel-headers, do not know. I can compile misdn-queue but misdn-start (after scan + config) tells me my module has an invalid format. Thanks Giorgio Incantalupo Kai Ober wrote: gcc -v : gcc version 4.1.0 no problems using latest stuff from

[asterisk-users] working chan_bluetooth enviroment

Hello, can anyone tell me which of the thousand chan_bluetooth development is the latest one and where can I get it? Which version of Asterisk is supposed to be the best choice with bt-support? Thanks alot Matthias ___ --Bandwidth and Colocation

[asterisk-users] Handytone 286 T.38 SDP parameters

Hello, I installed trunk version of asterisk. I'm testing T.38 fax. My configuration is FaxMachine1--HandyTone286--asterisk--spa2100--FaxMachine2. When I send fax from FaxMachine1, I cannot see any T.38 SDP parameters. Any idea is appreciated. Thanks. Jason.

Re: [asterisk-users] does misdn-mqueue work if compiled with gcc 4?

Giorgio Incantalupo schrieb: Hi Kai, thanks, maybe I used the wrong kernel-headers, do not know. Did you change them? Now everything works fine? regards ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To

Re: [asterisk-users] transform bridged call into a conference

you're searching for 3pty... which TECH do ya use? zap/misdn/bristuff/sip/ or do ya look for a generel solution? zap does this by itself! there is a possibility do throw calle 2 into an conference, get calle 3 throw it into conference, and them self join the conference. Klaus Darilion

[asterisk-users] Cisco 7970

Is 8.0.2.SR1 still the latest firmware? I still haven't managed to do anything useful with that weary expensive phone. It still only receives and places calls, nothing else. Is there any exciting feature that can work with asterisk and SIP firmware? Has anybody managed to do anything of the

Re: [asterisk-users] Asterisk t38passthrough

Finally it's working! I was doing everything well, the problem was that neither the latest branch of Asterisk-t38 worked (http://svn.digium.com/svn/asterisk/team/group/t38passthrough/), neither the patched version of Asterisk 1.2.7. Only the branch of Asterisk-t38 made from source version

Re: [asterisk-users] transform bridged call into a conference

Kai Ober wrote: you're searching for 3pty... which TECH do ya use? zap/misdn/bristuff/sip/ or do ya look for a generel solution? yes - general zap does this by itself! how? there is a possibility do throw calle 2 into an conference, get calle 3 throw it into conference, and them self

Re: [asterisk-users] Handytone 286 T.38 SDP parameters

Dear Jason, Only version 1.2.4 of the Asterisk-t38 branch worked for me. Matybe your problem could be that. Try to install version 1.2.4, it should work. Regards, Ricardo. Jason Kim wrote: Hello, I installed trunk version of asterisk. I'm testing T.38 fax. My configuration is

Re: [asterisk-users] Handytone 286 T.38 SDP parameters

Also try to use Kapanga softphones (www.kapanga.net) that are able to send T.38 faxes. I've tested one Grandstream handytone 386 ATA device, and it doesn't work well with T.38. I only have been able to do T38 fax pass-through between two Kapanga softphones with Asterisk in the middle. You

Re: [asterisk-users] transform bridged call into a conference

zap does this by itself! how? threewaycalling=yes in zapata.conf there is a possibility do throw calle 2 into an conference, get calle 3 throw it into conference, and them self join the conference. ok - how does it work? with app_chanredirect? this was used to run in astersik 1.09...

Re: [asterisk-users] does misdn-mqueue work if compiled with gcc 4?

Hi Kai, the problem is to find the right kernelI used apt-get *install* kernel-headers-*`**uname* -r*`* -y but it seems not to be the right one...Even zaptel is not working: FATAL: Error inserting zaptel (/lib/modules/2.6.15-1-486/misc/zaptel.ko): Invalid module format I've always

[asterisk-users] MOR and MCC - billing solutions for Asterisk released

Hello, Kolmisoft: http://www.kolmisoft.com released new versions of MCC and MOR - Billing solutions for Asterisk PBX MOR - is new product, it's MCC v2. Rewritten to support MySQL and is based on Ruby on Rails. ChangeLog for MCC: DB: New fields: providers.enable_cid_prefix varchar

Re: [asterisk-users] IP phone with 2 ethernet jacks

Matt Birmingham wrote: Can you supply any information/links as to how you got this working? Sure. I followed the instructions here: http://www.voip-info.org/wiki/view/Asterisk+presence However, there was one important fact missing. By default the presence feature is turned off in the

[asterisk-users] Asterisk Development and Release Cycle

Do Asterisk team care about this anymore? Whole text can be found here: http://www.asterisk.org/developers/releasecycle -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148 SIP: [EMAIL PROTECTED] e-mail: tparcina#lama.hr http://www.lama.hr

Re: [asterisk-users] Can I increase DTMF sensitivity?

Steve Edwards wrote: I have a client that is complaining that they are having trouble with Asterisk not reliably recognizing DTMF from some phones. The calls are being delivered by ATT Qwest over PRI to te410p's with firmware that was upgraded about 10 months ago. Can I tweak the amplitude

[asterisk-users] Asterisk codec strangeness

Hi All, I have two peers (call then peerA and peerB) on my server, both can accept g711, g729 and g723. However, when peerA initiates a request, asterisk decides to transcode g729 into ulaw when peerB could very well use g729... This behavior isn't very scalable (transcoding is CPU

[asterisk-users] playback() breaks audio in zap-iax-iax-zap channel

all,i'm having a problem where a callee is unable to hear the caller,but the caller can hear the callee perfectly well.the call is processed as follows:1. caller dials into zap-1 via a pri line ( using chan_zap ) 2. zap-1 connects to iax-1 over an iax channel Dial(IAX2/[EMAIL PROTECTED])3. iax-1

Re: [asterisk-users] does misdn-mqueue work if compiled with gcc 4?

Giorgio Incantalupo schrieb: Hi Kai, the problem is to find the right kernelI used apt-get *install* kernel-headers-*`**uname* -r*`* -y so the only i can tell is: - my kernel is 2.6.16.13-4-default (meaning suse 10.1 default) - installed latest zap and libpri packages from asterisk-org -

[asterisk-users] Re: transform bridged call into a conference

Hi! I wonder if it is possible to transform a bridged call into a conference. E.g. phone 1 calls phone 2 (normal bridged call with Dial()). Further phone 3 wants to join? Is this possible? Can you please refer me the proper applications? What I am doing: First, redirect to bridged calls to

[asterisk-users] Re: Nokia E60/61/70 and SIP

In article [EMAIL PROTECTED], [EMAIL PROTECTED] says... qualify=yes Put in in the sip.conf file in the configuration section for the specific phones. I don't think he thought on that. -- Tomislav Parčina Lama Computers Split Stinice 12, 21000 Split Tel.: +385(21)495148 Mob.: +385(91)1212148

Re: [asterisk-users] Grabbing authenticated mailbox value from VoicemailMain()

Still stumped on this one. Am I going to have to scrap the idea of using the outgoing call function of VoicemailMain as a way to authenticate and process an outgoing call (i.e. should I be building something specific using VMAuthenticate) ? N. On Mon, 28 Aug 2006 09:56:17 -0400, sip wrote Is

[asterisk-users] Analyze core file prodeced after safe_asterisk crashh

Hi, I have a Call Center running with safe_asterisk script.When Asaterisk crash produce a core file but I don´t know how analyze it!Any ideas?? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or

Re: [asterisk-users] does misdn-mqueue work if compiled with gcc 4?

Hi Kai, it works!!! Not with gcc 4.1 but with 4.0! Giorgio Incantalupo. Kai Ober wrote: Giorgio Incantalupo schrieb: Hi Kai, the problem is to find the right kernelI used apt-get *install* kernel-headers-*`**uname* -r*`* -y so the only i can tell is: - my kernel is

Re: [asterisk-users] Asterisk 1.2.4 I hear other party's voice only when I speack need help

Hi Tommaso, have you tried to search for noise suppression? I remember some phone has a function to automatically suppress it so the caller does not hear anything and thinks the other party has hung up. Giorgio Incantalupo Tommaso Calosi wrote: I have this problem with Asterisk 1.2.4 I

Re: [asterisk-users] Asterisk Manager Interface Question

100ms On 8/28/06, Roi Stork [EMAIL PROTECTED] wrote: The version that I'm using is 1.2.7.1. What is the default value of writetimeout in manager.conf? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To

[asterisk-users] Detect if cell phone or users

Hi List, I have sprint pcs cellular service and I discoverd thatI am able to send a text message to a landline. If I do I get an SMS from the saying I sent a text message to a landline. Then the landline that I sent a text message to gets a call with my message (text to speach). I was

[asterisk-users] Connecting two asterisk servers

Hi friends,Thank you to all for your response and cooperation to me. I have a doubt.I have two asterisk servers and contains two public IPs. One * server is in Florida (USA) and second * server is in Delhi (India).1) Is it possbile to connect these two * servers?2) The person who is registered

[asterisk-users] Which BRI Card ?

I've been asked to comment on how the Digium B410P stacks up, but have not had any experience of this card. Can anyone comment on the strengths / weaknesses or compare it to any other BRI isdn card that is asterisk compatible ? On another note, would it be better to buy a TDM04B instead of 4

[asterisk-users] IP interface box for Meridian type digital phone

Hi, Anyone can point me to a product that would allow to connect Meridian type digital phone to an Asterisk PBX. I am looking for something like an ATA that you would connect the digital phone to and the ATA would attached to the IP network going to the Asterisk server. Thanks for any

Re: [Asterisk-Users] Extension Ring on Multiple Phones

This is a reply to a fairly old thread. My EXTEN string is meant to ring 3 phones (will increase to 12) thus: old: exten =_879677[67],1,Dial(SIP/120) ; works fine new: exten =_879677[67],1,Dial(SIP/120SIP/122SIP/124) I edit extensions.conf to the new line above, type 'reload' into the

Re: [asterisk-users] ISDN BRI, and Trixbox

On Tue, 2006-08-29 at 00:59 +0100, Nigel Godfrey wrote: Hi, I have 2 Billion BRI ISDN cards, and intend to set up a Trixbox server. I currently have a Asterisk 1.0 server which has been running for a couple of years using Billion BRI cards and Junghanns BRIstuff. Trixbox, if it works,

RE: [asterisk-users] IP interface box for Meridian type digital phone

The Citel SIP Handset Gateway http://www.citel.com allows you to connect Meridian phones (and others) to SIP services, such as Asterisk (and others). -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Andre Courchesne - Consultant Sent: 29 August 2006

[asterisk-users] Advice needed - asterisk Mitel 200SX

Greetings: I have a Mitel 200SX PBX right now that is experiencing some voicemail difficulties. The voicemail system is a SmoothOperator system - yep, thats right, genuine 1995 Dos-based ISA motherboard technology running our corp voicemail system. In a word, boring and quickly becoming

RE: [asterisk-users] IP interface box for Meridian type digital phone

Take a look at the citel handset gateway. The SIP one. It's an ATA for 24 nortel/meridian phones http://www.citel.com/products/handset_gateways/ david -Message d'origine- De : [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Andre Courchesne - Consultant Envoyé : 29 août 2006

Re: [asterisk-users] Connecting two asterisk servers

Please search the wiki first. Most of your questions you post can easily be found by doing a search. Put some effort into finding the answers to your questions first and on your own, and then if you still have questions, I'm sure everyone would be more than willing to help. On 8/29/06, Crazy Boy

[asterisk-users] DTMF between cisco and sipura going through asterisk

Hello all, we're having an issue with DTMFs being sent to Sipura's. Calls are originating from a Cisco AS5300 being sent to asterisk which in turn sends it to the Sipura. Connected to the Sipura is a legacy PBX (or actually shows the same problem with a cheap answering machine). The DTMFs

[Fwd: Re: [asterisk-users] Asterisk t38passthrough]

You can download the patch for t.38 passthrough from the URL: http://bugs.digium.com/file_download.php?file_id=9335type=bug Regards, Ricardo. Patrick wrote: On Tue, 2006-08-29 at 12:50 +0100, Ricardo Carvalho wrote: Finally it's working! I was doing everything well, the problem was

Re: [asterisk-users] Standard for transfer via IAX provider?

On 25 Aug 2006, at 17:42, Henry J. Cobb wrote: Is there any standard way to signal to an IAX provider that I want them to conference in another Asterisk box located elsewhere and then hand off the call to the remote center after a short period of three-way talk? My problem is that I don't

[asterisk-users] Copying a recording to a voice mail box

Hello,I am new to asterisk and have a very newbie question. I am try to implement a simple IVR solution that prompts users to say an item, record that then prompt for another item record that.. etc... Here is what I have so far. [custom-lbp]exten = s,1,Playback(LBPsayname)exten =

[asterisk-users] Copying a recording to a voice mail box

Hello,I am new to asterisk and have a very newbie question. I am try to implement a simple IVR solution that prompts users to say an item, record that then prompt for another item record that.. etc... Here is what I have so far. [custom-lbp]exten = s,1,Playback(LBPsayname)exten =

[asterisk-users] Re: [asterisk-biz] Asterisk Tools

Interesting, I was asking about the FCC dictating new (read stupid) methods of doing this which will cause higher communications systems. The end user will get a bigger bill in the future because folks are faking their CallerID/ANI today. A private investigator making a few calls or a call

[asterisk-users] GXP-2000 auf Betafirmware updaten?

Hi, currently I use version 1.1.0.16 for my GXP-2000 which works really fantastic. The only drawback I see is the addressbook. Is the firmware 1.1.1.9 stable enough to use the phone in normal environment? The webpage http://www.voip-info.org/wiki/view/GXP-2000 says that there it is possible to

[asterisk-users] Mix Monitor call quality

Hi trying to record calls using mixmonitor, but I'm having problems with call quality the call seems OK but then it drops frames with silence ( for less than 0.5 seconds) then call continues All I'm doing is bridging two zap channels and recording no transcoding or changes to the channels

[asterisk-users] SIP T1 timer and qualify=yes

I ran in the same issues as John Todd did some while ago: http://lists.digium.com/pipermail/asterisk-users/2005-November/129541.html I use qualify=yes to ping our internal SIP proxies for failover and therefore I have very low delays, e.g. Name/usernameHostDyn Nat ACL Port

[asterisk-users] compile problems with app_rxfax.c and asterisk 1.2.11

Hi All, Trying to add faxing to asterisk but get a compile error. Any ideas? Is it broken for Asterisk 1.2.11 or was it me again :-) I followed the instructions from here: http://www.asteriskguru.com/tutorials/spandsp.html Thanks in advance Phil gcc -shared -Xlinker -x -o app_page.so

Re: [asterisk-users] Asterisk Development and Release Cycle

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Tomislav Parčina wrote: Do Asterisk team care about this anymore? I don't know. Do you use Asterisk? That makes you part of the team. Have you tested the trunk version? Provided assistance testing out patches waiting for completion? Really,

Re: [asterisk-users] Analyze core file prodeced after safe_asterisk crashh

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 equis software wrote: Hi, I have a Call Center running with safe_asterisk script. When Asaterisk crash produce a core file but I don´t know how analyze it! Any ideas?? http://www.asterisk.org/doxygen/AstDebug.html - -- Cheers, Matt Riddell

Re: [asterisk-users] Detect if cell phone or users

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Dovid Bender wrote: Hi List, I have sprint pcs cellular service and I discoverd that I am able to send a text message to a landline. If I do I get an SMS from the saying I sent a text message to a landline. Then the landline that I sent a text

Re: [asterisk-users] Voicemail/Email Integration

[EMAIL PROTECTED] wrote: Is there a way to implement voicemail/email integration such that you could retrieve the voicemail with either the phone or email, but only have to delete the message once? You can try our voicemail client called Tycho, available for MacOS X, Linux and Windooze. You

Re: [Asterisk-Users] Extension Ring on Multiple Phones

Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote: This is a reply to a fairly old thread.My EXTEN string is meant to ring 3 phones (will increase to 12) thus: old: exten

Re: [asterisk-users] Detect if cell phone or users

How do you handle situations where a cellphone number has been ported to a land line/VoIP provide or vice versa? The phone number isn't a reliable indicator of provider or medium.-brandon On 8/29/06, Matt Riddell (IT) [EMAIL PROTECTED] wrote: -BEGIN PGP SIGNED MESSAGE-Hash: SHA1Dovid

Re: [asterisk-users] Detect if cell phone or users

I don't think there's any authority in North America that tells you whether a number is a cellular number. However, it's conceivable you could write a script to access information available on, for example, www.telcodata.us, and check the prefix-type for a given phone number. The prefix type

[asterisk-users] OT: Bandwidth calculations and PCI/PCIX/PCIE

I found this interesting but old white paper at Dell.com tech solutions and another one from INTEL. It compares bandwidth usage of a PCI, PCI-X, PCI-E in33/66/100/133mhz bus and different technologies that can saturate the bus. It helped me understand the bandwidth required for TDM

Re: [asterisk-users] Advice needed - asterisk Mitel 200SX

Fix the computer It worked for 10 years, it can be made to continue working The worst thing you can do is try and implement such a radical change without lots of testing, and more testing. Unless, of course, you are the owner of the company or want to go on a job search! JMO John Novack Ron

Re: [asterisk-users] Asterisk t38passthrough

Hi Ricardo, On a 1.2.4 with the T.38 patch, I tried both t38pt_udptl = yes t38pt_rtp = yes t38pt_tcp = yes and t38pt_udptl = yes t38pt_rtp = no t38pt_tcp = no but still got ...chan_sip.c:3716 process_sdp: Unknown SDP media type in offer: image 5144 UDPTL t38 Warnings I tried it on Kapanga

RE: [asterisk-users] Connecting two asterisk servers

In short, yes... The wiki (http://www.voip-info.org) has documentation on how to configure your servers, how to configure the dialplan, etcI don't mean to single you out mate, but has anyone else noticed an increase in the number of questions being asked that could have been answered

RE: [Asterisk-Users] Extension Ring on Multiple Phones

That's very very odd...that should work fine :( -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Larry Alkoff Sent: Tuesday, August 29, 2006 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users]

Re: [asterisk-users] Analyze core file prodeced after safe_asterisk crashh

It's not clear if the OP wanted 1) information on how to analyse the core file or 2) provide information to the bug tracker for others to analyse. Matt's answer addresses #2. How about #1? Anybody care to share their techniques for analysing a core dump? On Tue, 29 Aug 2006, Matt Riddell (IT)

Re: [asterisk-users] Connecting two asterisk servers

Crazy Boy a écrit : Hi friends, Thank you to all for your response and cooperation to me. I have a doubt. I have two asterisk servers and contains two public IPs. One * server is in Florida (USA) and second * server is in Delhi (India). 1) Is it possbile to connect these two * servers?

[asterisk-users] Asterisk 1.2.11 and ${SIPDOMAIN} variable

Hi, I've just upgraded from Asterisk 1.2.10 to 1.2.11 and I've noticed that the ${SIPDOMAIN} variable now contains a different (and to my mind, incorrect) value than what it used to. Instead of (say) example.com, it now contains the string example.com;user=phone instead which causes calls to

Re: [asterisk-users] Detect if cell phone or users

On 13:56, Tue 29 Aug 06, Jay Milk wrote: I don't think there's any authority in North America that tells you whether a number is a cellular number. However, it's conceivable you could write a script to access information available on, for example, www.telcodata.us, and check the

[asterisk-users] CPU configuration for 250 calls SIP to SIP to IAX and fonebridge and two asterisk servers

Hi, I would like to read your comments for the following setup: Building A: 3 voice E1incoming toa quad redfone fonebridge (TDMoE) The fonebridge goes to a port in a 24 port gigabit switch in the gigabit switch VLAN1 is for the fonebridge and the first gigabit NIC on a dual NIC server in the

Re: [asterisk-users] Analyze core file prodeced after safe_asterisk crashh

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Steve Edwards wrote: It's not clear if the OP wanted 1) information on how to analyse the core file or 2) provide information to the bug tracker for others to analyse. Matt's answer addresses #2. How about #1? Anybody care to share their

Re: [asterisk-users] Asterisk codec strangeness

Jean-Michel Hiver a écrit : Hi All, I have two peers (call then peerA and peerB) on my server, both can accept g711, g729 and g723. However, when peerA initiates a request, asterisk decides to transcode g729 into ulaw when peerB could very well use g729... This behavior isn't very

Re: [asterisk-users] does misdn-mqueue work if compiled with gcc 4?

On Tue, Aug 29, 2006 at 02:49:54PM +0200, Giorgio Incantalupo wrote: Hi Kai, the problem is to find the right kernelI used apt-get *install* kernel-headers-*`**uname* -r*`* -y but it seems not to be the right one...Even zaptel is not working: FATAL: Error inserting zaptel

[asterisk-users] Detecting sound before answer

Title: Message Hello, Lets say Im dialing out and before channels are bridged I hear beep or something similar. That way I know Im calling to other Telco/Provider. Is it possible to detect that beep before channel is answered and to redial through other trunk?

[asterisk-users] Administrator Forum Email

Hi List; Can someone advise me what is the email of the administrator forum so I can send for him to fix my account? The forum that I am talking about it existed in the following link: http://forums.digium.com/ Regards Bilal Ghayad __ Do You

Re: [asterisk-users] Does anyone use T.38?

We have successfully used Sipura 2100 ATAs for this with an external fax machine connected to its FXS port. The Sipura is connected to a Cisco fax gateway right now, we haven't been able to test it with Asterisk yet. On Fri August 25 2006 06:58, Ricardo Carvalho wrote: Does anyone use T.38

RE: [asterisk-users] Does anyone use T.38?

XMedius is a great T.38 fax product, integrate with LDAP/AD/Exchange. Integrates with the PRI card in our Cisco Routers using H.323. - Disclaimer: This e-mail communication and any attachments may contain confidential and privileged information and is

Re: [asterisk-users] Asterisk Development and Release Cycle

- Tomislav Parčina [EMAIL PROTECTED] wrote: Do Asterisk team care about this anymore? Whole text can be found here: http://www.asterisk.org/developers/releasecycle Of course we care. Turns out that schedule was unrealistic, and when we start the next cycle we will regroup and decide if

Re: [asterisk-users] Asterisk codec strangeness

are your codec allow= statements in the same order in each peer block? meaning does peerA have g729 at a different priority than peerB? Moj Jean-Michel Hiver wrote: Jean-Michel Hiver a écrit : Hi All, I have two peers (call then peerA and peerB) on my server, both can accept g711, g729 and

Re: [asterisk-users] Copying a recording to a voice mail box

There are flags to the VoiceMail application that instruct it to behave differently than normal. It probably won't let you append the messages into one message, however. That seems like it would be a problem in this project, but if it's not, you might try: VoiceMail([EMAIL PROTECTED])

Re: [asterisk-users] Asterisk codec strangeness

Mojo with Horan Company, LLC a écrit : are your codec allow= statements in the same order in each peer block? meaning does peerA have g729 at a different priority than peerB? Aah, thanks that fixed it because most of the traffic is g729. Now, if peerA does send me ulaw instead of g729

[asterisk-users] Digium makes the list!

There is a link on Groklaw for the following article: Open source companies to watch Digium makes the second entry on the list. Link below: http://www.networkworld.com/news/2006/082806-open-source.html?ts Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase

Re: [Asterisk-Users] Extension Ring on Multiple Phones

Color me puzzled. What part of: exten = _879677[67],1,Dial(SIP/120) should be deleted? Larry William Piper wrote: Sounds like you still have the old exten still there. Make sure you get rid of: exten = _879677[67],1,Dial(SIP/120) bp On 8/29/06, Larry Alkoff [EMAIL PROTECTED] wrote: This

[asterisk-users] New Parrot application, repeats what you say and more!

Lobster Technologies has just anounced the release of the most annoying open source IVR application ever devised by lobsters called PhoneParrot. PhoneParrot is an app that uses silence detection to repeat everything a person says in to the phone. http://www.lobstertech.com/code/phoneparrot/

Re: [asterisk-users] Asterisk codec strangeness

I suspect so, but I'm not sure :) Jean-Michel Hiver wrote: Mojo with Horan Company, LLC a écrit : are your codec allow= statements in the same order in each peer block? meaning does peerA have g729 at a different priority than peerB? Aah, thanks that fixed it because most of the traffic is

RE: [Asterisk-Users] Extension Ring on Multiple Phones

Then entire OLD extension must be removed so the new one will match -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Larry Alkoff Sent: Tuesday, August 29, 2006 6:49 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Re: [Asterisk-Users] Extension Ring on Multiple Phones

The whole thing. Both (old and new)have the same exten and the same priority, you can't do that and expect it to work properly. The new extenwill call all 3 phones at the same time, whoever answers first gets the call. If you want it to callSIP/120 first and if they don't answer then ring to all

Re: [asterisk-users] Asterisk Performance without RTP?

On Tue, 29 Aug 2006, Nick Hoffman wrote: On Tue August 29 2006 04:39, Greg Boehnlein [EMAIL PROTECTED] wrote: On Mon, 28 Aug 2006, Andrew Kohlsmith wrote: On Monday 28 August 2006 13:02, Greg Boehnlein wrote: I've pushed over 1,000 concurrent calls this way using the SIPP program

Re: [asterisk-users] Re: [asterisk-biz] Asterisk Tools

On 8/29/06, Andrew Latham [EMAIL PROTECTED] wrote: Interesting, I was asking about the FCC dictating new (read stupid) methods of doing this which will cause higher communications systems. The end user will get a bigger bill in the future because folks are faking their CallerID/ANI today. A

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