[asterisk-users] Dialplan matching

Hello all, I am trying to figure out the logic in on prefix matching for Asterisk 1.4.5. I want to be able to pass all international calls EXCEPT calls to 011870, 01137455 and so on. exten = _011870.,1,Goto(intl-disabled,s,1) exten = _01137455.,2,Goto(intl-disabled,s,1) exten =

Re: [asterisk-users] BLF in Asterisk 1.4.*

-10-29 5:15 AM, Asterisk User an.asterisk.u...@gmail.com wrote: Hello everybody, does anybody know if BLF is correctly working in Asterisk 1.4.*? I'm particularly interested in Asterisk 1.4.25. Thanks in advance! Phil

[asterisk-users] BLF in Asterisk 1.4.*

Hello everybody, does anybody know if BLF is correctly working in Asterisk 1.4.*? I'm particularly interested in Asterisk 1.4.25. Thanks in advance! Phil -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Delay on sip channel

Hi, My SIP service provider terminates calls in meetme in my Asterisk PBX and am getting delay on those channels. I found following link to measure delay in meetme and to decrease it eventually. http://lists.digium.com/pipermail/asterisk-dev/2005-August/014958.html It says, enable USE_RTC for

[asterisk-users] DUNDILOOKUP doesn't return record

Hi All, Found an issue with DUNDILOOKUP function in Asterisk 1.6.0.5. I was using DUNDIQUERY (Set(ID=${DUNDIQUERY(${MNUM},priv,b)})) for dundilookup and it was working fine. But when I tried to use DUNDILOOKUP function (Set(DL=${DUNDILOOKUP(${MNUM},priv,b)})), it didn't retuen me a result.

Re: [asterisk-users] Codec translation in Asterisk

Nobody to take this one! Am I missing anything in knowing following issue? --Hi Group, --Can anybody explain me in detail how the codec translation happens on --asterisk side when 2 endpoints have different codecs? --Thanking you in advance. SM --

[asterisk-users] Codec translation in Asterisk

Hi Group, Can anybody explain me in detail how the codec translation happens on asterisk side when 2 endpoints have different codecs? Thanking you in advance. --SM -- _ -- Bandwidth and Colocation Provided by

Re: [asterisk-users] help on ${RTPAUDIOQOS}

it looks like it has something to do with the way a call is hungup. Has anybody else any idea? Thanks, ---Asterisk User ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register

[asterisk-users] help on ${RTPAUDIOQOS}

-- Packet2Packet bridging SIP/555-b7a80948 and SIP/666-089cb090 -- Executing [...@incoming_vpbx:1] NoOp(SIP/555-b7a80948, ***) in new stack Thanking you... ---Asterisk User ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Help sending call to local server

or iax.conf and it works as calls get landed in particular context of remote server. Would you please suggest me changes to be made in .conf file(s) if I want the calls to be landed in context of local server if Server_ip is the IP of a server running asterisk? Thanking you --ASTERISK USER

[asterisk-users] Help setting IAX variables.

of SA? If yes then how? I know about IAXVAR application where variables set in source server of IAX channel can be access from destination server... Any help is greatly appreciated. ---Asterisk User ___ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] Help setting IAX variables.

for your inputs. --- Asterisk user ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options

[asterisk-users] Voicemail retention

Asterisk version 1.2.27 We are running into issues where people are not deleting their voicemails and it is filling up the storage for voicemail. We would like to run a script that dumps all voicemail that are older than X days. Can we simply check the date time stamp on the message

[asterisk-users] Show call coming back from Call Parking

Our operator has asked if it is possible that when a call times out in the call parking and comes back to her, if there is someway to show that call has come back from parking. I have looked all over the documentation and have come up with nothing so far. All I see when a call times out is:

Re: [asterisk-users] unable to call ATT audio conference bridge

Hello, Can someone help me with this please? Attached is the log file. thank you Original Message Subject:[Fwd: asterisk-users Digest, Vol 26, Issue 166] Date: Fri, 29 Sep 2006 10:31:21 -0400 From: asterisk-user [EMAIL PROTECTED] To: asterisk-users

[asterisk-users] Asterisk Directory listing

How do I take out few extensions (vm enabled extensions) from the default company directory listing? thanks. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] [Fwd: asterisk-users Digest, Vol 26, Issue 166]

; charset=ISO-8859-1; format=flowed On 9/28/06, asterisk-user [EMAIL PROTECTED] wrote: Hello, I have a problem with asterisk and trying to see if someone can help me fix the issue... Problem: I couldn't join ATT's Tele Conference bridge directly without their customer service interaction. Instead

[asterisk-users] unable to call ATT audio conference bridge

Hello, I have a problem with asterisk and trying to see if someone can help me fix the issue... Problem: I couldn't join ATT's Tele Conference bridge directly without their customer service interaction. Instead of getting the automated prompts to join the conference, it takes me to the

[Asterisk-Users] QSIG suopprt in Asterisk

I am trying to use QSIG to interoperate with legacy PBXs. I am looking to see whether any one knows whether Call Hold, Call Transfer, MWI works with QSIG support in Asterisk. Thanks in advance. --dp ___ --Bandwidth and Colocation provided by

Re: [Asterisk-Users] QSIG support in Asterisk

I am trying to use QSIG to interoperate with legacy PBXs. I am looking to see whether any one knows whether Call Hold, Call Transfer, MWI works with QSIG support in Asterisk. Thanks in advance. --Pillai On 5/4/06, Olivier Krief [EMAIL PROTECTED] wrote: 2006/5/3, Marco Mouta [EMAIL

[Asterisk-Users] QSIG support in Asterisk

I am looking to get the info about QSIG support in Asterisk. Does Asterisk have QSIG support? Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking? If so, How to configure that? Thanks ___ --Bandwidth and Colocation provided by

[Asterisk-Users] QSIG support in Asterisk

I am looking to get the info about QSIG support in Asterisk. Does Asterisk have QSIG support? Does Asterisk support QSIG SIP Tunneling or QSIG SIP Interworking? If so, How to configure that? Thanks --dp ___ --Bandwidth and Colocation provided by

Re: [Asterisk-Users] VoiceMail realtime not working in asterisk-1.2.6

hi, odbc show is printing like Name: asterisk DSN: asterisk Connected: yes with regrads asteriskusers --- Nathan Bowyer [EMAIL PROTECTED] wrote: On 4/4/06, asterisk user [EMAIL PROTECTED] wrote: hi all, I can not get voicemail working in realtime with asterisk-1.2.6. extconfig.conf

[Asterisk-Users] VoiceMail realtime not working in asterisk-1.2.6

hi all, I can not get voicemail working in realtime with asterisk-1.2.6. extconfig.conf is correct voicemail = odbc,asterisk,voicemail_users i am getting the fallowing error Executing Answer(SIP/xx.xx.xx.xxx-0a02e1c0, ) in new stack -- Executing Set(SIP/xx.xx.xxx-0a02e1c0, foo=102) in new

[Asterisk-Users] need an bench-marking tool

hi, i wanted to test a conference, so can any one help me in finding out a bench-marking tool in which we can set different codecs for each user. with regards vicky ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users

[Asterisk-Users] ZAP - Can't pickup calls on Analog Trunk

Are the trunks just pots lines (plain old telephone service lines)? If you don't know you could put an analog phone on the incoming lines and verify you can dial out. Also, if you call the line the phone should ring. If this is true then you will need fxoks in your pbx instead of the fxsks.

[Asterisk-Users] Call Parking - Set ID on return

We have just analog lines coming in to our Asterisk box and so no CallerID information can be gathered, all calls look the same on the phone display. Once a user parks a call and the time runs out it returns the call but keeps the original CallerID information that makes it look like it is just

[Asterisk-Users] ZAP - Can't pickup calls on Analog Trunk

We have 4 analog line and 2 analog trunks. On the trunks we have all the DIDs coming into the current phone system. Trying to get everything moved over to Asterisk but having issues picking up the calls on the analog trunk. We can receive calls on the plain analog lines and we can call out on

RE: [Asterisk-Users] Recommendations on a WiFi phone for *?

Has anyone tried out Hitachi IPC-5000 ? It looks nice and it's a bit exensive, but I would still like to hear how does it behave around Asterisk. Ivan ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To

[Asterisk-Users] Sipura + Asterisk 1.2 + dtmf

Hi, I have a problem with Sipura and Asterisk 1.2... everything was working smoothly with 1.0.9 until I upgraded to 1.2. The DTMF tones are no longer working, I cannot access Voicemail or send DTMF digits anywhere. What changed in version 1.2?? I've read many people with the same issue but with

RE: [Asterisk-Users] Sipura + Asterisk 1.2 + dtmf

of updated firmware. On 12/12/05, Asterisk User [EMAIL PROTECTED] wrote: Hi, I have a problem with Sipura and Asterisk 1.2... everything was working smoothly with 1.0.9 until I upgraded to 1.2. The DTMF tones are no longer working, I cannot access Voicemail or send DTMF digits anywhere. What changed

RE: [Asterisk-Users] Sipura + Asterisk 1.2 + dtmf

+INFO. and in sip.conf I have rfc2833 On 12/12/05, Asterisk User [EMAIL PROTECTED] wrote: Hi, I have a problem with Sipura and Asterisk 1.2... everything was working smoothly with 1.0.9 until I upgraded to 1.2. The DTMF tones are no longer working, I cannot access Voicemail or send DTMF digits

[Asterisk-Users] Re: problem with registration of SIP phone

, Asterisk User [EMAIL PROTECTED] wrote: I'm runing [EMAIL PROTECTED] beta6 and I have a I'm runing [EMAIL PROTECTED] beta6 and I have a problem with registration of SIP phone. I can't find/replicate when exactly its happends but sometimes after server restart or phone restart or after long idle

[Asterisk-Users] Re: problem with registration of SIP phone

, Asterisk User [EMAIL PROTECTED] wrote: I managed to isolate the problem a bit more, maybe it will help to find a solution:The problem with the phones is not the initial registration, but the re-registration process.When I create a new extension the phone registers ok, but when the same phone tries

[Asterisk-Users] SIP Registration Problem

I'm runing [EMAIL PROTECTED] beta6 and I have a problem with registration of SIP phone. I can't find/replicate when exactly its happends but sometimes after server restart or phone restart one of the phone can't register and I get this in the server: Transmitting (no NAT) to

[Asterisk-Users] problem with registration of SIP phone

I'm runing [EMAIL PROTECTED] beta6 and I have a I'm runing [EMAIL PROTECTED] beta6 and I have a problem with registration of SIP phone. I can't find/replicate when exactly its happends but sometimes after server restart or phone restart or after long idle time the phones can't register and I get

[Asterisk-Users] cmd dial timeout don't work in asterisk 1.2 ?

Hello. My dial timeout worked perfectly on the last asterisk but not on the new. Here is my extension.conf : exten = s,1,Answer() exten = s,2,noop(${CALLERID}) exten = s,3,Set(TIMEOUT(response)=20) exten = s,4,Background(test) exten = s,5,Dial(Zap/2|${CALLERID},15) exten =

Re: [Asterisk-Users] cmd dial timeout don't work in asterisk

asterisk user dupont wrote: Hello. My dial timeout worked perfectly on the last asterisk but not on the new. Here is my extension.conf : exten = s,1,Answer() exten = s,2,noop(${CALLERID}) exten = s,3,Set(TIMEOUT(response)=20) exten = s,4,Background(test) exten = s,5,Dial(Zap/2

[Asterisk-Users] In France asterisk never detect hang up. Why ?

Hello. I am sorry my english is not good at all. When i have a call from a fxo port of a tdm400p, asterisk waits one minute before detecting that the caller has hang up his phone. I have in my extension conf : answer background (the prompt is 40 second long) dial (on fxs port) confgured for

[Asterisk-Users] Re: Asterisk en france

Bonjour, J'ai changé en tel que ci dessous, et j'ai toujours le même probleme. Il detect toujours pas le raccroché. I have changed to this new file, and i still have the same problem. Still not detecting hang up. [channels] language=fr default=fr relaxdtmf=yes rxgain=0.0 txgain=0.0

RE: [Asterisk-Users] WiFi Phones

We bought a couple of the UTStarCom phones. They work fine in the office environment where noise is low, but on our production floor it is impossible for me to hear what is being said and the person on the other end of the call also says that they cannot hear a thing from the F1000 when the

Re: [Asterisk-Users] Cisco 7960 / SIP tftp configs

Thanks for the responses. All is happy. For the record the correct answers are: Q1 - Additions/changes SIPxx.cnf take effect on reboots, deletions do not. A1 - Don't just comment out the line setting, change it specifically to UNPROVISIONED. Q2 - How to get Message button working. A2 -

[Asterisk-Users] Cisco 7960 / SIP tftp configs

I have three questions about my 7960 phone that I can't discern from the docs/wiki. 1st - If I change the SIPxx.cnf file to change registrations it sets up new lines as expected. If I delete a line it doesn't get removed when I reboot the phone. I have to go to the phone, unlock it, and

RE: [Asterisk-Users] Wengo config and G729(a)

Hello Here you go : [wengo-outgoing] type=peer fromuser= username username= username secret=password host=voip.wengo.fr fromdomain=voip.wengo.fr disallow=all allow=alaw allow=ulaw dtmfmode=inband canreinvite=yes nat=yes insecure=very dtmf=inband context=wengo-outgoing authname= username This is

[Asterisk-Users] Asterisk Realtime - How to enable the debug message for SIP users query

Hi experts, I wish someone would kindly give me a hand on a problem on Asterisk Realtime. May I know how to enable the debug messages for the Asterisk SIP Registrar query the SIP user data in the created MySQL table. I found that I can see the debug message for cdr_mysql which shows it can

[Asterisk-Users] Asterisk SIP cannot restrict call from softphone before registration

Hi all, I have problem with my Asterisk. I'm using the softphone Xten-Lite.I've removed the SIP client information in sip.conf. The softphone can't register to Asterisk, but it can make outgoing calls. I've tried to add back the SIP client information into the sip.conf, but make a wrong

[Asterisk-Users] FXO-FXS parameters

Hello, I'm trying to get feed back from other Asterisk users of Welltech WellGate 3701A / 3702A Or Micronet SP5012s / SP5014s Or Immix Tel C3-FXS/FXO Or Euro Teletech VIP-400 (All those are in fact the same product...) Trying to find/share ideas/comments about registrations issue, caller ID

[Asterisk-Users] Wellgate 3701

Hi everyone I'm trying to setup this Welltech Wellgate 3701 box. If I got to the proxy setup it seems to work but the Pstn incoming call always got a voice prompt from the Wellgate. Going to peer to peer mode seems to be better but I couldn't find any working configuration inside Asterisk. I

[Asterisk-Users] Voicemail as email attachment not working individually i.e. extensions specific

with if attach=no is set globally then attach=yes will not work if set for some particular extension. I have googled for it but couldnt found anything useful.please help.. Thanks Asterisk-user __ Do you Yahoo!? Yahoo! Mail - now with 250MB free storage

[Asterisk-Users] Zultys ZIP2

Hello All, I'm having trouble getting a Zultys ZIP2 to work with Asterisk, along with some other troubles in general. I keep getting a Got SIP response 481 Call Leg/Transaction Does Not Exist back from x.x.x.x). Even when Asterisk reports that the ZIP2 registered correctly, I can't make any

[Asterisk-Users] incoming caller doesn't hear rining.

Hi, I have an asterisk installation that has been happily working in production for some time (E100P and UK BT ISDN30). Recently I upgraded to HEAD-07/29/04. Now, incoming callers don't hear ringing while calling in. As far as I can tell, my config files haven't changed from what was working

[Asterisk-Users] (no subject)

Hi All, I recently upgraded from a very old stable to HEAD. For some reason, incoming callers don't hear ring tones when calling in. Everything else is working fine. Where should I look for a fix? ISDN -- X100P -- asterisk -- sipphones. Thanks Johan

[Asterisk-Users] no incoming pstn ring tone

Hi All, I recently upgraded from a very old stable to HEAD. For some reason, incoming callers don't hear ring tones when calling in. Everything else is working fine. Where should I look for a fix? ISDN -- E100P -- asterisk -- sipphones. Thanks Johan

[Asterisk-Users] Windows Messenger Problem

Hi All, I'm currently having a problem with video calls with 2 Windows Messenger clients through *. The video+audio call gets established ok, but after a random period, usually anywhere from 20 seconds to 3 minutes, (though usually under a minute), both video and audio gets disconnected. Just

Re: [Asterisk-Users] Video/H323/SIP

Florian Overkamp wrote: Hi, -Original Message- Is there any software based solution to establish a video connection with * and sip protocol? MSN messenger 4.7 with any windows capturing device should work. Make sure you force the codecs properly, because MSN tries to negotiate

Re: [Asterisk-Users] Using Windows Messenger+Video in *

Florian Overkamp wrote: Hi, -Original Message- Has anybody used Windows Messenger with asterisk? All documents around (google - wiki - bugs.digium.com) say that asterisk supports windows messenger with video but i have no succes yet! I can establish connection with audio but no

[Asterisk-Users] VoiceXML support and integration

Hi All, Do any of you know what the status is for VoiceXML support in * ? Is it already existing, or is it planned for the future? If it's not in now, do you know on what type of scale the work would be to integrate VXML into * ? Thanks in advance

[Asterisk-Users] No read routine on channel AsyncGoto/Zap/1-1ZOMBIE

I saw the error: No read routine on channel AsyncGoto/Zap/1-1ZOMBIE in my log today. Despite googling, I have no idea what this error relates to. Could someone please help me. Thanks JC ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] Asterisk and Multicast

Hi, Could anyone tell me if asterisk supports multicast? And if so, what type? And if not, are there any plans to implement one in the forseeable future? Thanks, Jason ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] CVS Changes (NAT-SIP)

I had been running an older patched CVS to get VOIP working with NAT and everything had been running fine. I just built * on a new box with CVS-01/18/04-12:19:25. And now I can get remote SIP users to register. Has anything major changed... [general] port = 5060 ; Port to