[asterisk-users] Runaway MOH/mp3123 process?

Has anyone noticed a problem with runaway mpg123 processes for music-on-hold eating up ~100% CPU and driving the load on the machine way up? I've seen this problem consistently with multiple Asterisk installs, 1.2.x and 1.4.x, although admittedly it was more common with 1.2.x as far as I can t

Re: [asterisk-users] using Playback() to play a random sound file

From: Steve Edwards <[EMAIL PROTECTED]> Date: Tue, 1 May 2007 22:08:10 -0700 (PDT) On Tue, 1 May 2007, Yuan LIU wrote: From: Steve Edwards <[EMAIL PROTECTED]> Date: Tue, 1 May 2007 21:10:40 -0700 (PDT) On Tue, 1 May 2007, Jay Austad wrote: I've got a directory under /var/lib/asterisk/sounds w

Re: [asterisk-users] Zaptel kernel module load order

On Tue, May 01, 2007 at 02:53:33PM -0700, C. Chad Wallace wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Mitch Jackson wrote: > > Evening, > > > > My latest asterisk box is having a difficult problem. It is > > configured with one TE210P and TDM400P with four FXO modules. I'm > > r

Re: [asterisk-users] Improving Asterisk's DNS support

On Wed, 2007-05-02 at 01:34 -0400, Brian Capouch wrote: > Kristian Kielhofner wrote: > > > > > The intent of my original message was to try to start a discussion > > on how we can fix a REAL, KNOWN problem with Asterisk to make it > > better. I'm not sure what the exact problems are (or even th

RE: [asterisk-users] Applet?

Yeh I know but it's just frustrating me, this is the third time that when someone has asked about web Click to Talk solutions Salvatore has piped up with Estara because he used to manage a hotel PABX where they used it. And that's all cool - it's a public list, but it's a $50,000 solution not $2,0

Re: [asterisk-users] Applet?

Pablo L. Arturi wrote: > Hello people. I would like to know if someone knows about any applet to > include in a web page to start calls. What I am looking for is something > that doesn't allow users to change numbers, or any other option, so I > can include it in my web page and force them to call

Re: [asterisk-users] Digital Phones

Salvatore Giudice wrote: > Nortel digital Meridian phones are like $400/each. At least that was the > price of the phones at a hotel I did a job for recently. Still? (Is Nortel even making these phones anymore? I thought they spun off their telephone set division -- anybody heard of Aastra? ;) )

Re: [asterisk-users] Improving Asterisk's DNS support

Kristian Kielhofner wrote: The intent of my original message was to try to start a discussion on how we can fix a REAL, KNOWN problem with Asterisk to make it better. I'm not sure what the exact problems are (or even the specific symptoms in all cases), but I am willing to offer anything tha

Re: [asterisk-users] Applet?

Dean Collins wrote: > Dude yes we know Estara is so cheap you said this 2 months ago, you said > this last month and you are saying this today. > > Yet every customer that comes to us to buy a license says their quote > was around $50,000 for the first year for Estara click to talk. > > I’m prepa

Re: [asterisk-users] using Playback() to play a random sound file

On Tue, 1 May 2007, Yuan LIU wrote: From: Steve Edwards <[EMAIL PROTECTED]> Date: Tue, 1 May 2007 21:10:40 -0700 (PDT) On Tue, 1 May 2007, Jay Austad wrote: I've got a directory under /var/lib/asterisk/sounds which contains a bunch of sound files. I would like to call the Playback command to

Re: [asterisk-users] using Playback() to play a random sound file

From: Steve Edwards <[EMAIL PROTECTED]> Date: Tue, 1 May 2007 21:10:40 -0700 (PDT) On Tue, 1 May 2007, Jay Austad wrote: I've got a directory under /var/lib/asterisk/sounds which contains a bunch of sound files. I would like to call the Playback command to play the files, but I need it to sel

Re: [asterisk-users] using Playback() to play a random sound file

On Tue, 1 May 2007, Jay Austad wrote: I've got a directory under /var/lib/asterisk/sounds which contains a bunch of sound files. I would like to call the Playback command to play the files, but I need it to select a file to play randomly. Is there any way to do this? I do this with an AGI.

[asterisk-users] using Playback() to play a random sound file

I've got a directory under /var/lib/asterisk/sounds which contains a bunch of sound files. I would like to call the Playback command to play the files, but I need it to select a file to play randomly. Is there any way to do this? ~jay ___ --Band

Re: [asterisk-users] Trixbox 2 and MFC/R2

On 4/23/07, Carlos Chavez <[EMAIL PROTECTED]> wrote: Can anyone recommend which versions of spandsp, libsupertone, libunicall and libmfcr2 to use to install Unicall on a Trixbox 2.0 machine? [...] Carlos, Use the latest snapshots for Asterisk 1.2 . They are working pretty well. Let

Re: [asterisk-users] Display Caller ID of called party

Quoting "Savoy, Kevin - Williston, ND" <[EMAIL PROTECTED]>: Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just diale

Re: [asterisk-users] Autoattendant press 1 collides with extension numbers...

On 5/1/07, Stephen Bosch <[EMAIL PROTECTED]> wrote: J. Oquendo wrote: > So I have whose autoattendant is colliding with their extensions... > Quick fix anyone? > > Second someone presses say a person's extension (101) ... Autoattendant > sends them to the first context... Two things: 1 -- your

Re: [asterisk-users] Digital Phones

You can also look at adsi phones, they should work well with a channel bank. Aastra makes some quite nice ADSI phones. They are not digital but can do a lot that digital phones could. On 5/1/07, bilal ghayyad <[EMAIL PROTECTED]> wrote: Hi List; Asterisk does not have any kind of cards that can

RE: [asterisk-users] Digital Phones

You can use something like this which supposedly works well and is easy to configure. Costs about $120 per port full retail and works with all sort of phones including Nortel. http://www.citel.com/Products/Portico.asp -Original Message- From: Dean Collins [mailto:[EMAIL PROTECTED] Se

RE: [asterisk-users] Applet?

Dude yes we know Estara is so cheap you said this 2 months ago, you said this last month and you are saying this today. Yet every customer that comes to us to buy a license says their quote was around $50,000 for the first year for Estara click to talk. I'm prepared to send a $100 bottle

RE: [asterisk-users] Applet?

You must be stoned. I have a client that has Estara service and only paid $2k USD for it. I’m not going to ‘show’ you anything. If you want to see what BNN has, you need to call BNN. They do have a web-based applet. It’s based on Windows RTC. They have a lot more behind the counter than thei

RE: [asterisk-users] Digital Phones

Nortel digital Meridian phones are like $400/each. At least that was the price of the phones at a hotel I did a job for recently. When you go to SIP, you may save on the capital costs for the phones, but other costs will increase. These are related to: 1.) increased support requirements for suppo

RE: [asterisk-users] Applet?

Salvatore, As I've said before Estara charge $US50,000 minimum price for their solution. We discussed this last time, I'm still waiting for you to show me a Click-to-Talk solution for anywhere near our price of £1,100 per server for the Corraleta SDK. Secondly I went to Bluenotenetworks, th

RE: [asterisk-users] How many users can be supported simultaneously?

I would test that under load and see what kind of QoS you get. If you have mixed traffic streaming and non-streaming coming from the server, you will need 802.1p and an 802.1p compatible router. If possible, you may want to add a second wireless interface, bind asterisk to that interface, and have

RE: [asterisk-users] Applet?

If you want a commercial service, there are some decent companies out there like Estara. http://www.estara.com/ These services don't come cheap. Bluenote Networks also has a web CTC applet for their SIP PBX that they license to enterprises. http://www.bluenotenetworks.com They have a pretty ni

Re: [asterisk-users] Improving Asterisk's DNS support

On 5/1/07, Per Jessen <[EMAIL PROTECTED]> wrote: Kristian Kielhofner wrote: > After several years of using Asterisk I have always been frustrated > by the support for DNS. I have seen all kinds of strange behavior > when Asterisk is used on a system with "iffy" DNS servers: Maybe that's where

RE: [asterisk-users] Applet?

Hi Pablo, no not off topic but probably better to be asked in the -biz list. To answer your question.there is either Jiax which is free or there is our company www.Mexuar.com with the Corraleta SDK, which is fully customizable, fully brandable and able to be use

RE: [asterisk-users] Digital Phones

What brand of digital phones, I think I read some time ago that someone was doing something with Nortel phones but I seem to remember the cost of the phone meant...better to toss the handsets and buy new sip handset. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph

Re: [asterisk-users] TC400B

bilal ghayyad wrote: Hi List; When I need TC400B? When you need 96 channels of G729. If I have a solution of 4 users (IP Phones) and 4 analoge lines, then I need TC400B? NO, just buy 4 G729 licenses. Regards Bilal __ Do You Yahoo!? Ti

RE: [asterisk-users] Display Caller ID of called party

Alex wrote: > It seems to me that what you are really talking about > is manipulating the display features of the phone. > Caller ID is unlikely to have this effect as the phone > does not consider the From: URI in the SIP header unless > the call is of an incoming nature. This feature is ofte

RE: [asterisk-users] Re: Wildcard TDM11B & Wildcard TDM04B

From: bilal ghayyad <[EMAIL PROTECTED]> Date: Tue, 1 May 2007 14:56:14 -0700 (PDT) Hi Noah; ut TDM11B contains physically 4 ports, if it supports only 1 FXS and 1 FXO, then what shall we do in the other two ports already existed? You can populate two more interface modules. Yuan Liu Regards

RE: [asterisk-users] How many users can be supported simultaneously?

Thank you for the reply..VPN is used for remote access and for secure data transfer..The web-server does not have a lot of traffic..I use SIP and there is a grandstream gateway with 4 FXO..I think that 25 calls is a good number.. Από: [EMAIL PROTECTED] εκ μέρους

[asterisk-users] TC400B

Hi List; When I need TC400B? If I have a solution of 4 users (IP Phones) and 4 analoge lines, then I need TC400B? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com _

[asterisk-users] Applet?

Hello people. I would like to know if someone knows about any applet to include in a web page to start calls. What I am looking for is something that doesn't allow users to change numbers, or any other option, so I can include it in my web page and force them to call to me and no one else. I ha

[asterisk-users] Digital Phones

Hi List; Asterisk does not have any kind of cards that can work with it to be used with Digital Phones (digital phones differ than analoge phone and differ than IP Phones). Anyone can advise about this as I did not find this on Diguim Regards Bilal Ghayad ___

[asterisk-users] Re: is dundi worth pursuing in this situation?

At work, I have 4 branch offices at which I've deployed asterisk. Call termination/origination at each branch office is handled either through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing the legacy PBX at our HQ with asterisk. Each branch office has between 3 and 20 employees,

[asterisk-users] Re: Wildcard TDM11B & Wildcard TDM04B

Hi Noah; ut TDM11B contains physically 4 ports, if it supports only 1 FXS and 1 FXO, then what shall we do in the other two ports already existed? Regards Bilal __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around htt

Re: [asterisk-users] Stanaphone business ok?

Not sure, Talk to tech support and get the info. You can always just go to another provider and transfer your DID. -bkruse Todd H wrote: I see that stanaphone is not accepting new customers. Does anyone know if they are doing ok? I have a number with them and would like to start redirecti

Re: [asterisk-users] Zaptel kernel module load order

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Mitch Jackson wrote: > Evening, > > My latest asterisk box is having a difficult problem. It is > configured with one TE210P and TDM400P with four FXO modules. I'm > running FC6. > > The TE210P only has a single PRI. > > When the system boots, it is

Re: [asterisk-users] Display Caller ID of called party

On 5/1/07, Savoy, Kevin - Williston, ND <[EMAIL PROTECTED]> wrote: Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I ju

Re: [asterisk-users] Display Caller ID of called party

Kevin, It seems to me that what you are really talking about is manipulating the display features of the phone. Caller ID is unlikely to have this effect as the phone does not consider the From: URI in the SIP header unless the call is of an incoming nature. The solution to this is bound to

RE: [asterisk-users] Stanaphone business ok?

Hmmm that's not good, I've been very happy using them as a backup line to my packet 8 services. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Todd H

RE: [asterisk-users] Stanaphone business ok?

Write them and ask. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -Original Message- From: [EMAI

Re: [asterisk-users] Delay in Dial()

Eric "ManxPower" Wieling wrote: > Stephen Bosch wrote: >> Eric "ManxPower" Wieling wrote: >>> Steve Finkelstein wrote: All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP ph

RE: [asterisk-users] Calls in ulaw, not gsm as desired

How did you set it to 5053? Can you post your sip.conf? You should remove the passwords and ip addresses, etc. Usually, it's just an allow and a disallow statement inserted into each inbound and outbound channel definition. -- Salvatore Gi

Re: [asterisk-users] is dundi worth pursuing in this situation?

Salvatore Giudice wrote: > DUndi or enum only make sense if you plan to move extentions dynamically > without having to touch you Asterisk configs or if you want to expose your > addressing to the outside world. > > Personally, I would do it statically so you can avoid delays in processing > addre

RE: [asterisk-users] OT: Capture Asterisk traffic

Ethereal will let you export an rtp stream as a .au file. That's one of the very minor items we cover in our conference series and our VoIP 100 course. There is a lot more fun to be had when you get into RTP sequence number prediction and RTP stream I injection. --

RE: [asterisk-users] T1 interface

You could get yourself a cisco universal gateway or a Audiocodes Mediant 1000 Single Span T1 SIP Gateway. With regard to the cards: In my experience, you want an echo cancellation card if you are connected to a carrier without echo cancellers. Typically, LEC circuits do not have echo cancellers an

Re: [asterisk-users] MYSQL application in dial plan

Yehavi Bourvine +972-8-9489444 wrote: > Hello, > > I would like to implement a few decision making process inside the dialplan > using information stored in MySQL (like LCR, etc.). I see the MYSQL() > application, but as far as I understand I have to connect to the database each > time I want to

Re: [asterisk-users] Delay in Dial()

It's amazing how simple some answer are. Thank you kindly for your responses Edoardo and Luki. :-) - sf Edoardo Serra wrote: > Hi Steve, >put a timeout in the Dial command, if the call isn't answered it > returns after the timeout has expired > > e.g.: > exten => _X.,1,Dial(SIP/${EXTEN}|15)

[asterisk-users] Re: [asterisk-dev] SRTP implementation

Olle E Johansson wrote: 23 apr 2007 kl. 19.55 skrev Russell Bryant: John Todd wrote: To morph this into a -dev thread: if this patch were to become (again) useful and error-free, is there any objection or usefulness in adding it to TRUNK? Personally, I think there is, if there is a method b

[asterisk-users] Display Caller ID of called party

Not sure if this can be done or not, but I can't seem to find it anywhere on the Wiki. When dialing interoffice with Asterisk 1.4.2, I would like to have the caller id of the person I am dialing displayed and not the number I just dialed. Is this possible? So, if extension 4023 is John Doe, and I d

[asterisk-users] Stanaphone business ok?

I see that stanaphone is not accepting new customers. Does anyone know if they are doing ok? I have a number with them and would like to start redirection people before it gets canceled on me if they are having trouble thanks Todd ___

Re: [asterisk-users] Delay in Dial()

Stephen Bosch wrote: Eric "ManxPower" Wieling wrote: Steve Finkelstein wrote: All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is se

[asterisk-users] chan_sip seems to be hanging

I posted about this problem last week and thought it was a combination of SIP/ZAP causing issues in Asterisk. Since then I've realized it's only the SIP channel that's hanging. When this happens a call can still come in and hit the IVR, but no one can connect to the server from a SIP client.

Re: [asterisk-users] Calls in ulaw, not gsm as desired

That's what I did though. So my sip.conf file no longer has any "allows" in it. Instead, it should be relying on the realtime settings for that. However, even though I told it to only use 5053, it still is using ulaw. Rob Salvatore Giudice wrote: > > Yeah that is fine. You don't need to do any mo

Re: [asterisk-users] Improving Asterisk's DNS support

I haven't been using Asterisk for long, but I have not yet experienced any DNS-related oddities. Then keep using it, and you will. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options

Re: [asterisk-users] is dundi worth pursuing in this situation?

On 5/1/07, Bruce Reeves <[EMAIL PROTECTED]> wrote: Your setup is very similar to one of my own, and I started of manually configuring it, creating IAX connections for each site and then using dial plan to route the call. When I looked at Dundi and finally got it working, I have one IAX connectio

Re: [asterisk-users] OT: Capture Asterisk traffic

I remember an app called 'vomit' that could allegedly reconstruct audio files from tcpdump pcap files. Salvatore Giudice wrote: I think you want: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp dst portrange 5060-65534 dst port port True if the packet is ip/tcp, ip/udp, ip6/tcp or ip

[asterisk-users] T1 interface

Would anyone care to recommend a T1 interface method for Asterisk that would function as an (external) alternative to a PCI card like the Digium TE120P? Like some sort of T1-SIP gateway? Also, would anyone with experience using these products care to comment on the practical value of the TE207

Re: [asterisk-users] MYSQL application in dial plan

Well, you should be able to leave it open. However, I don't know what would happen if MySQL times out and disconnects the connection because it considers it stale. I don't know if you can check that error and reconnect. Yehavi Bourvine +972-8-9489444 wrote: Hello, I would like to implement

Re: [asterisk-users] Delay in Dial()

Eric "ManxPower" Wieling wrote: > Steve Finkelstein wrote: >> All, >> >> Is there any syntax I can use to put a delay in two lines being dialed? >> One is a SIP endpoint, the other is my cell phone. I'd like to have the >> SIP phone ring for some arbitrary number of seconds before it is sent >> off

RE: [asterisk-users] My Sip Provider lacks Sip 2.0 183 (Ringing)information

From: Knud Müller <[EMAIL PROTECTED]> Date: Tue, 01 May 2007 15:19:17 +0200 Hi all, my sip provider does'nt send a 183 Message when the opposite party rings. It sends the ringing indication on the audio stream. Is there any chance that the asterisk can analyze this audio stream (meta) informa

Re: [asterisk-users] OT: Capture Asterisk traffic

CSB wrote: > I want to capture all my Asterisk traffic (including RTP) and then > analyse it. > > My plan was to use tcpdump and then analyse with Wireshark. The > following works: > tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 > > But I want to be a bit more selective: > tcpdump -C 100 -W 10 -w /tmp/t

Re: [asterisk-users] OT: Capture Asterisk traffic

wireshark can further filter out what you don't want, you can also pipe the dump to "grep" and match only what you want On May 1, 2007, at 11:32 AM, CSB wrote: I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wire

Re: [asterisk-users] is dundi worth pursuing in this situation?

DUNDi would be very well suited to this particular application. Publish the extensions that are reachable at each location and when one site dials an extension it gets routed to the one that says "i have this". ENUM would probably work just as well for this. I like ENUM with PowerDNS and MYSQL

RE: [asterisk-users] is dundi worth pursuing in this situation?

DUndi or enum only make sense if you plan to move extentions dynamically without having to touch you Asterisk configs or if you want to expose your addressing to the outside world. Personally, I would do it statically so you can avoid delays in processing addressing especially - in the case of enu

Re: [asterisk-users] TDM400P and TE405P

On Tue, May 01, 2007 at 02:01:33PM -0400, Nitesh Divecha wrote: > Hello All, > > To avoid conflicts I removed TE405P and left the TDM400P and > reconfigured the card using "genzaptelconf". > > When I run ztcfg -vv I saw the card and modules are loaded and also I > used "ztmonitor 1 -v" and I sa

Re: [asterisk-users] OT: Capture Asterisk traffic

The RTP traffic is not going to be on port 5060, that is the sip only. Check your rtp.conf file in asterisk for the port range used for RTP traffic. On 5/1/07, CSB <[EMAIL PROTECTED]> wrote: I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdu

RE: [asterisk-users] Calls in ulaw, not gsm as desired

Reload will reload your sip.conf file! As well as iax.conf, extensions.conf, queues.conf, voicemail.conf, users.conf From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Rob Schall Sent: Tuesday, May 01, 2007 2:06 PM To: Asterisk Users Mailing List - Non-Commercial Discus

RE: [asterisk-users] OT: Capture Asterisk traffic

I think you want: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp dst portrange 5060-65534 dst port port True if the packet is ip/tcp, ip/udp, ip6/tcp or ip6/udp and has a destination port value of port. The port can be a number or a name used in /etc/services (see tcp(4P) and udp(4P)).

Re: [asterisk-users] did we all get spammed by TechnoCo ?

Me 3. - Original Message - From: "Salvatore Giudice" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Tuesday, May 01, 2007 5:38 PM Subject: RE: [asterisk-users] did we all get spammed by TechnoCo ? That stuff is so dangerous. There are too m

RE: [asterisk-users] Calls in ulaw, not gsm as desired

Yeah that is fine. You don't need to do any more than that. -- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-29

[asterisk-users] MYSQL application in dial plan

Hello, I would like to implement a few decision making process inside the dialplan using information stored in MySQL (like LCR, etc.). I see the MYSQL() application, but as far as I understand I have to connect to the database each time I want to query it; this seems a CPU eater to me. Is this i

Re: [asterisk-users] is dundi worth pursuing in this situation?

Erik, Your setup is very similar to one of my own, and I started of manually configuring it, creating IAX connections for each site and then using dial plan to route the call. When I looked at Dundi and finally got it working, I have one IAX connection for all sites and the connections are dynami

Re: [asterisk-users] is dundi worth pursuing in this situation?

I have run into the exact same situation and have the same question. I did it in the dial plan manually due to time contraints but if DUNDi or ENUM or something else is better suited I would love to know. Also the guides and tutorial that I found did not touch on specifics for a situation like t

Re: [asterisk-users] Channel stuck with call pri flag

You mean a PRI debug trace? right now I have some channels that are in this state. There is not much I can do as this is a production system... John Treble wrote: Can you get a layer 3 analyzer trace (e.g., Q.931) and post it here? John Treble Ottawa, Ontario, Canada -Original Messa

RE : [asterisk-users] How do I do this in Asterisk?

Hi Christian, Increase a variable in the menu loop, or exactly in the "t" and "i" extensions like this : exten => s,1,Wait(3) exten => s,n,Answer() exten => s,n,Set(LoopStep=1) exten => s,n,Set(TIMEOUT(digit)=3) exten => s,n,Set(TIMEOUT(response)=10) exten => s,n,Wait(1) exten => s,n(menuresta

RE: [asterisk-users] Channel stuck with call pri flag

Can you get a layer 3 analyzer trace (e.g., Q.931) and post it here? John Treble Ottawa, Ontario, Canada > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Andre Courchesne - Consultant > Sent: May 1, 2007 12:44 PM > To: asterisk-us

Re: [asterisk-users] Calls in ulaw, not gsm as desired

I was in the asterisk console and I typed "reload". Is this not enough to reload the sip.conf file? Rob Andreas Sikkema wrote: >> However, even once I reloaded the extensions, its still only >> using ulaw. >> > > You didn't reload the sip config? Maybe that's your problem? > > _

Re: [asterisk-users] Voicemail Creation

You can use real time with an agi. - Original Message - From: mohammad mirzaee To: asterisk-users@lists.digium.com Sent: Sunday, April 29, 2007 12:50 PM Subject: [asterisk-users] Voicemail Creation HI All; I want to use Asterisk for just Voicemail Server and I need a Dyn

Re: [asterisk-users] Test

Test emails and out of office emails make my day. - Original Message - From: "Wilson Pickett" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, May 01, 2007 5:37 PM Subject: Re: [asterisk-users] Test where are the out of office replies

Re: [asterisk-users] TDM400P and TE405P

Hello All, To avoid conflicts I removed TE405P and left the TDM400P and reconfigured the card using "genzaptelconf". When I run ztcfg -vv I saw the card and modules are loaded and also I used "ztmonitor 1 -v" and I saw the gain moving up and down. I did create trunks and outbound routes usin

[asterisk-users] OT: Capture Asterisk traffic

I want to capture all my Asterisk traffic (including RTP) and then analyse it. My plan was to use tcpdump and then analyse with Wireshark. The following works: tcpdump -i eth0 -s 0 -w /tmp/tcpdump.1 But I want to be a bit more selective: tcpdump -C 100 -W 10 -w /tmp/tcpdump -i eth1 -s 0 udp a

Re: [asterisk-users] Delay in Dial()

Steve Finkelstein wrote: All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait

RE: [asterisk-users] Calls in ulaw, not gsm as desired

> However, even once I reloaded the extensions, its still only > using ulaw. You didn't reload the sip config? Maybe that's your problem? -- Andreas Sikkema ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNS

[asterisk-users] is dundi worth pursuing in this situation?

At work, I have 4 branch offices at which I've deployed asterisk. Call termination/origination at each branch office is handled either through a frac PRI or 3rd party SIP provider. Soon, I'll be replacing the legacy PBX at our HQ with asterisk. Each branch office has between 3 and 20 employees,

[asterisk-users] Channel stuck with call pri flag

Hi, I have a problem where some PRI channels get stuck in a "Call" mode. If I do a zap show channel XX, it shows as "PRI Flags: Call". However there is no calls on that channel. Trying to force a hangup does not work: [EMAIL PROTECTED] Dialer]# asterisk -r -x "soft hangup zap/27-1"

Re: [asterisk-users] Delay in Dial()

Hi Steve, put a timeout in the Dial command, if the call isn't answered it returns after the timeout has expired e.g.: exten => _X.,1,Dial(SIP/${EXTEN}|15) It waits 15 secs for the call to be answered Look at http://www.voip-info.org/wiki-Asterisk+cmd+Dial for more informations Regards

RE: [asterisk-users] Calls in ulaw, not gsm as desired

It's probably not your codec. Do you have your asterisk box on a Voice VLAN with priority queing configured? If you have mixed traffic on your uplink without VLAN's and priority queuing (or possibly 802.1p), then your QoS will suffer. Changing your codec to GSM will lower bandwidth consumption, but

Re: [asterisk-users] How many users can be supported simultaneously?

Antonopoulos Angelos wrote: > I have a pc with the following characteristics: > > Pentium IV 2.4Ghz HyperThreading > 512 MB PC3600 Dual DDR RAM > Seagate 80GB SATA HDD > 4-port ethernet 10/100 PCI Card > Netgear MA-311 802.11b Wireless Card > On this machine runs a VPN server, an Apache server

Re: [asterisk-users] Delay in Dial()

Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. Not directly, but yes. Hint: Local channel + Wait. Something like this: Dial(SIP/phone&Local/[EMAIL PROTECTED]) [delayed] exten => XX,1,Wait(10) exten => X

[asterisk-users] How do I do this in Asterisk?

Hi all, I have created a menu from which the caller can select several options such as being transfered to our phones and my mobile phone, meetme, etc. If the caller press an invalid option i have set it to play a message like invalid choice please try again. If the caller make three invalid cho

RE: [asterisk-users] How many users can be supported simultaneously?

Will you be allowing reinvites? If the server processes media, it will obviously support less simultaneous calls. Also, you may want to rethink the wireless portion. Odds are you will have horrible QoS problems if you run multiple calls or mixed traffic over wireless. BTW, what do you use VPN for?

RE: [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help

Hi All, I tried to send this email this morning, but I think it has been moderated due to size issue's, so I'll resend it again in 3 parts: PART 2 Database Table Definition (taken from asterisk readme's) CREATE FUNCTION loin (cstring) RETURNS lo AS 'oidin' LANGUAGE internal IMMUTABLE

RE: [asterisk-users] Asterisk 1.4.4 VoiceMail ODBC Storage Help

Hi All, I tried to send this email this morning, but I think it has been moderated due to size issue's, so I'll resend it again in 3 parts: PART 1 Hi All, Just an update, after looking a little further into this, it appears that * tries to delete a record that does not exist before inserting it

RE: [asterisk-users] How many users can be supported simultaneously?

The answer is about 42 handsets... Seriously though - you don't mention traffic on the vpn server, you don't mention traffic on the apache server you don't mention anything about transcoding, conference rooms, or if you are using SIP or IAX. You ask an unanswerable question so my ans

RE: [asterisk-users] Cisco 7940 no outgoing audio

You should get a packet capture of both cisco-cisco and grandstream/polycom-cisco. Compare the SDP's. The cisco phone may not be able to understand the other vendor's devices. BTW, what version of firmware are you running on the cisco phones? -- Salv

[asterisk-users] Calls in ulaw, not gsm as desired

My problem is this We have a location outside of our network which is done over vpn. Everything works except for the voice quality to that location isn't very good. To try to resolve this, I wanted to try to make all calls go over gsm. Right now, when i say "show sip channels", they all show u

[asterisk-users] Delay in Dial()

All, Is there any syntax I can use to put a delay in two lines being dialed? One is a SIP endpoint, the other is my cell phone. I'd like to have the SIP phone ring for some arbitrary number of seconds before it is sent off to the mobile phone. Using something like a Wait() within a Dial() would be

[asterisk-users] Email to HP Product Suggestions - Seamless Transparent Fax Gateway

i sent a product suggestion to HP. It was a request to use software that already exists in their JetDirect and Multifunction Fax machines to make them seamlessly interoperate with a fax gateway in a way transparent to the end user. Essentially, giving the sysadmin a choice in fax transport mech

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