Re: [asterisk-users] Route an incoming call by ANI*DNIS

Dan Casey wrote: Sorry for my very delayed response. To answer a few questions: 1. Right, the *ANI*DNIS* is not working correctly. When the telco sends it, we are always missing the beginning of it. I almost always get a 7 digit ani, but sometimes it is 8 or 6. I won't be able to help with

[asterisk-users] Problems with losing D-Channel on

Hello all, I got a problem at an asterisk server, with dropping calls, losing all channels and reaktivating all channels and beeing back up. This problem seems to occure randomly over the whole day, when it gots traffic on the card. After looking @ google I found several hints but none did work

[asterisk-users] sl75 wlan not able of being pickuped?

Hello. I have a strange problem. Its not possible to pickup a call that was placed with a Siemens SL75 Wlan. When this phone calls an internal number and i try to pickup (*8) the call from my phone i get nothing. It seems i have the call for one second or so but after that the call is being

Re: [asterisk-users] California based PSTN connections

Just a follow up, I have my server with Cari.net in San Diego. How do you go about getting a block of DIDs and performing my own origination? Anyone has any experience in this field? Thanks. On 11/19/07, Eric Chamberlain [EMAIL PROTECTED] wrote: We use VoicePulse Connect. They now have a POP

[asterisk-users] Realtime - mysql query gives wrong results??

Hi, I am using Realtime for sip configuration. When there is an INVITE which arrives at asterisk asterisk makes the following selects: Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine. [Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:138

Re: [asterisk-users] channels to destroy

Hello, On Nov/19/2007, Johansson Olle E wrote: 16 nov 2007 kl. 14.06 skrev Carles Pina i Estany: In a couple of Asterisks, after type sip show channels we have a lot of these: IP_PEER dst_number something00102/00103 unkn No (d) Rx: BYE IP_PEER dst_number2 something2

[asterisk-users] MediaHandling--Help Required

Hello Users, My Setup is like this openser --Registrar asterisk --Callflow using asterisk-b2bua + radius for accounting My Intention was to generate a Acct-Stop Packet when there is a failure of RTP media from one of the UAC's( callee or caller) who is in dialog. so that the Caller will

[asterisk-users] SMS Feature In Asterisk

Is Asterisk capable of sending text messages to a cell phone or is there an application for that? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] SMS Feature In Asterisk

broadband Voice wrote: Is Asterisk capable of sending text messages to a cell phone or is there an application for that? Yes. Any carrier that supports SMS over analog lines will work with the Asterisk SMS application. Generally carriers in the USA and Canada do not support SMS over analog

[asterisk-users] not sending bye

Hello, We are using this Asterisk: 1.2.14-BRIstuffed-0.3.0-PRE-1y Everything works fine but we have an issue (not often, but one call every some hundreds) I sniffed the communication between phone, Asterisk and softswitch. I can see that Asterisk receives a Cancel from phone but Asterisk never

Re: [asterisk-users] SMS Feature In Asterisk

On Nov 20, 2007 6:24 AM, Eric ManxPower Wieling wrote: [...] but do generally have an e-mail-SMS gateway. Check with your carrier. http://en.wikipedia.org/wiki/SMS_gateways -- ___ --Bandwidth and Colocation Provided by

[asterisk-users] store 2 separate records in cdr when a call is transferd

Hi i've read this post http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html I just want to know if there are some upgrades... on 1.4 or 1.2. I'd like to store two records in the CDR instead of one, when a call is transferd. Is it possibile now? Thanks to all --

Re: [asterisk-users] store 2 separate records in cdr when a call is transferd

nik600 wrote: Hi i've read this post http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html I just want to know if there are some upgrades... on 1.4 or 1.2. I'd like to store two records in the CDR instead of one, when a call is transferd. Is it possibile now? Thanks

Re: [asterisk-users] r2 multiframe error - solved

All errors was genereted by physical link. Protocolvariant cz,10,6 its ok for me in Poland Thanks for help Regards Akron ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update

Re: [asterisk-users] How to enable res_config_mysql

okay, probably a typing issue check in extconfig.conf you have a line that is something like sipusers = mysql.asterisk_4,some_table_blablabla and it should be sipusers = mysql,asterisk_4,some_table_blablabla Note the change from period to comma right after mysql. Otherwise post that

Re: [asterisk-users] Realtime - mysql query gives wrong results??

Two things: 1. Set the context 2. Set the port Hi, I am using Realtime for sip configuration. When there is an INVITE which arrives at asterisk asterisk makes the following selects: Nov 20 10:37:10] DEBUG[31852]: res_config_mysql.c:651 mysql_reconnect: MySQL RealTime: Everything is fine.

Re: [asterisk-users] store 2 separate records in cdr when a call is transferd

for blind transfer! Many thanks! On Nov 20, 2007 2:24 PM, Atis Lezdins [EMAIL PROTECTED] wrote: nik600 wrote: Hi i've read this post http://lists.digium.com/pipermail/asterisk-dev/2007-May/027666.html I just want to know if there are some upgrades... on 1.4 or 1.2. I'd like to

[asterisk-users] OT - What is Alarm receiver feature ?

Hello, From http://www.asterisk.org/support/features or http://www.voip-info.org/wiki/index.php?page=Asterisk%20Features , there is a features list I'm trying to translate and explain to prospective customers. I can't relate this Alarm receiver feature to anything meaningful. Does it mean

Re: [asterisk-users] OT - What is Alarm receiver feature ?

Quoting Olivier [EMAIL PROTECTED]: if you are going to be a security company that receives alarm notification from burglar/fire alarms, this is the module for you - otherwise ignore it. Hello, From http://www.asterisk.org/support/features or

Re: [asterisk-users] Pass CallerID when call forwards to PSTN?

On Nov 20, 2007 5:06 AM, Dovid B [EMAIL PROTECTED] wrote: From what I have seen in the past asterisk should pass along the CID automatically. As some one else already mentioned. It can be your ITSP. You can always set the CID with Set(CALLERID(num)=1234567890). Asterisk does pass the caller ID

[asterisk-users] Realtime extensions configuration - calling user filtering

Hi, Is it possible to filter the calling user with the usage of mysql realtime the same as it is done in extensions.conf file: exten = some_exten/calling-user is there some flag which activates this extra check?? Cheers Tomasz ___ --Bandwidth and

[asterisk-users] ACD functionality , Skills for agents

Hi all, I have a question regarding ACD for queues. What happens when I have 2 or more queues with same weight and each queue has a call in? How will it decide which call will be routed to the next available agent? Will it take the call with the longest waiting time in queue? If not how

Re: [asterisk-users] ACD functionality , Skills for agents

On Nov 20, 2007 10:16 AM, Kyriakos [EMAIL PROTECTED] wrote: I have a question regarding ACD for queues. What happens when I have 2 or more queues with same weight and each queue has a call in? How will it decide which call will be routed to the next available agent? Will it take the

Re: [asterisk-users] Realtime extensions configuration - calling user filtering

As much I as can tell, Asterisk version 1.2 doesn't support the ex-girlfriend logic that you ask. I didn't test that feature with 1.4 releases, maybe they already implement it. Regards, Ricardo Carvalho.. On Nov 20, 2007 2:51 PM, Tomasz Zieleniewski [EMAIL PROTECTED] wrote: Hi, Is it

Re: [asterisk-users] Route an incoming call by ANI*DNIS

On Tuesday 20 November 2007 02:38:38 Atis Lezdins wrote: I won't be able to help with hardware part, but there's a simple trick to get them as you want: [incoming] _X.,1,Set(DNIS=${CUT(${EXTEN:-4})}) _X.,2,Goto,dnis,${DNIS},1 [dnis] 6789 = ... I don't think you've actually tested this,

Re: [asterisk-users] How to integrate Asterisk with Avaya

which Avaya system? and what are you trying to add with asterisk? On 11/20/07, Dovid B [EMAIL PROTECTED] wrote: Hello Everyone, Can someone please point to sources how to integrate Asterisk PBX with Avaya..? What normalize and expose protocol/API does Avaya support which can be use

Re: [asterisk-users] r2 multiframe error - solved

Jakub Syrek wrote: All errors was genereted by physical link. Protocolvariant cz,10,6 its ok for me in Poland Thanks for help Regards Akron Thanks. I will make a note of that in the code. Steve ___ --Bandwidth and Colocation Provided by

Re: [asterisk-users] Pass CallerID when call forwards to PSTN?

On Tuesday 20 November 2007 08:50:06 Russell Horn wrote: On Nov 20, 2007 5:06 AM, Dovid B [EMAIL PROTECTED] wrote: From what I have seen in the past asterisk should pass along the CID automatically. As some one else already mentioned. It can be your ITSP. You can always set the CID with

Re: [asterisk-users] r2 multiframe error - solved

Good news. On Nov 20, 2007 7:51 AM, Jakub Syrek [EMAIL PROTECTED] wrote: All errors was genereted by physical link. Protocolvariant cz,10,6 its ok for me in Poland Thanks for help Regards Akron ___ --Bandwidth and Colocation Provided by

Re: [asterisk-users] How to integrate Asterisk with Avaya

Hi, Im not talking about connecting Asterisk with Avaya system. I just mentioned Avaya because on a presentation I've been to, they said that this could be done. I want to do this on Asterisk. I already have a call centre setup with 5 different queues with same weights but it seems that ACD is

Re: [asterisk-users] blind transfer dumping calls

On Mon, 2007-11-19 at 16:26 +0200, Atis Lezdins wrote: On 11/19/07, Brian J. Murrell [EMAIL PROTECTED] wrote: I am using asterisk 1.4.10 and seem to be having a problem with blind transfer. This could very well be a pebkac problem but I'm not sure. This is probably issue with 1.4.10. I

Re: [asterisk-users] How to integrate Asterisk with Avaya

Please ignore last message of mine. I was busy with doing multiple tasks here at work and I falsely thought this was a reply to a mail I sent just a while ago. :P -Original Message- From: Kyriakos [mailto:[EMAIL PROTECTED] Sent: Tuesday, November 20, 2007 5:42 PM To: 'Asterisk Users

Re: [asterisk-users] Realtime extensions configuration - calling user filtering

I tried it with 1.4 and it didn't work with standard settings and no magic:) On Nov 20, 2007 4:32 PM, Ricardo Carvalho [EMAIL PROTECTED] wrote: As much I as can tell, Asterisk version 1.2 doesn't support the ex-girlfriend logic that you ask. I didn't test that feature with 1.4 releases, maybe

Re: [asterisk-users] Realtime extensions configuration - calling user filtering

If you really want to use some DB to read/write your dialplan, the best thing for you would be to write some scripts to generate text files from the contents of the tables of your DB. Those files can then be loaded in the extensions.conf file with the sentence: #include generated_file.txt. In the

[asterisk-users] Reporting bugs

I recently subscribed to the bugs mailing list and submitted a suspected bug. The report seems to be ignored. I am guessing that it is being ignored because I am not actually an asterisk user and I am unable to supply the version or configuration of the suspect site. So then I thought I should

Re: [asterisk-users] Help: How to configure SIP domain on SPA942

Take a look at the admin guides at http://spc.pifiu.com On Nov 18, 2007 10:53 PM, Philip Prindeville [EMAIL PROTECTED] wrote: I'm using a bunch of SPA942's, and I'm trying to provision them mostly by DHCP (and what I can't set that way, I try to provision via HTTP interface into the phone).

Re: [asterisk-users] Reporting bugs

Hi On Tue, Nov 20, 2007 at 09:33:59AM -0800, Robert Dyck wrote: I recently subscribed to the bugs mailing list and submitted a suspected bug. I figure you refer to http://lists.digium.com/mailman/listinfo/asterisk-bugs . This list is not used by users to report bugs. It is used by the

Re: [asterisk-users] Reporting bugs

Robert Dyck wrote: Two questions. How to get a report of a suspected bug to be taken seriously? http://bugs.digium.com How to get an account for one of the forums? That I don't know, I've never used them. Doug -- Ben Franklin quote: Those who would give up Essential Liberty to

[asterisk-users] FXO Hangs up automatically

Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas? I've searched the archives without much success. I appreciate all your

Re: [asterisk-users] [asterisk-dev] trunk working under windows!

Cool, i'll help out a bit with the windows port, i will start right away with a new project on asteriskguru making nightly executable builds and installers - will post the links in -users when i'm done. Well done luigi, this will make it a lot easier for a lot of non linux guys to make their

[asterisk-users] e911

One of my providers has a different SIP account for each number. I have all of my users in one outbound context (caller ID passes fine). How do I ensure that the callers get routed down their correct SIP account with my provider for e911 purposes without each having their own context? -

Re: [asterisk-users] Help: How to configure SIP domain on SPA942

Yeah, I looked at LinksysSPATFTPProv.pdf... It doesn't say, however, how to get the phone's configuration out as a flat XML file. Only how to push the file back into the phone. Nor does it say how the phone derives its SIP domain. -Philip [EMAIL PROTECTED] wrote: Take a look at the admin

[asterisk-users] iaxpeers from Realtime

Hello asterisk users, here is a little problem pulling out iax peers from real time database I have the following peer configured in my database mysql select name,username,secret,type,context,host,disallow,allow,defaultip,deny,permit, ipaddr,port from iax_users where name='iaxtermination';

Re: [asterisk-users] FXO Hangs up automatically

On Tue, Nov 20, 2007 at 09:01:22PM +0300, Timothy Smith wrote: Hi, I'm having problems using a TDM400P Card. I put my SIM card in a Nokia Premicell and connected it to a TDM400P card but when I make calls to the number, it hangs up automatically. The card also can't call out. Any ideas?

[asterisk-users] automatic blind transfer calls

Hi, I would like to do a blind transfer in an automatic way. For example I dial 5 during a call and the caller is blind transferred to SIP/578 (for example). I saw that with features.conf it is not possible to do that. Regards gianrico

Re: [asterisk-users] Route an incoming call by ANI*DNIS

Thank you all, It just so turns out that it was a bad zaptel module. We saw another post on digiums site where someone was having the exact same problem with several versions of zaptel. We changed to the one that he said worked (1.2.21), and all is well now. (And asterisk is now parsing the

[asterisk-users] Bugtracker to use with Asterisk?

Hello Now that I have my first IVR up and running, I'd like to have Asterisk create tickets in a bug tracker every time a call comes in. It's a nice way to know who's calling and why, before following up on the cause for the call. There are tons of bugtracking apps out there. Do you know of some

Re: [asterisk-users] Interface with NEC NEAX 2400

Is there anyone out there who has tried to connect up an asterisk box to make and take calls through a NEC NEAX 2400 using Q.sig or anything like it? Can anyone tell me if it is possible? Phil, I've successfully connected my NEAX 2400 to Asterisk using line side and trunk side T1's. I've

Re: [asterisk-users] Realtime extensions configuration - calling user filtering

On Tuesday 20 November 2007 11:18:45 Ricardo Carvalho wrote: If you really want to use some DB to read/write your dialplan, the best thing for you would be to write some scripts to generate text files from the contents of the tables of your DB. Those files can then be loaded in the

[asterisk-users] Cisco phones and 32 directory object limit

Hello List, For those of you with Cisco phones and XML directories and large user bases, how do you handle the 32 directory object limit? I know you can create multiple xml files with 32 objects in each but this just seems really sloppy. I would like to have one large directory. Thanks

Re: [asterisk-users] [asterisk-dev] trunk working under windows!

but ... why? Zoa wrote: Cool, i'll help out a bit with the windows port, i will start right away with a new project on asteriskguru making nightly executable builds and installers - will post the links in -users when i'm done. Well done luigi, this will make it a lot easier for a

[asterisk-users] FXO incomming call hangup problem

Dear all I have asterisk with TDM808B FXO port with i call in asterisk and i promt IVR then user dial extention for user then my SIP phone rining but i disconnect or hangup my mobile phone but still SIP phone rining and stop rining after timeout is there any PSTN problme

[asterisk-users] How to receive manager events from commands made by an AGI script?

Hi all, I'm new on this list, my name is Noel. :D I developed a system using AGI and now I'm trying to develop a system that listen events fired by Manager API. I have realized that I don't receive events from commands made by an AGI script like play a file or record a file. Is there a way to

Re: [asterisk-users] Asterisk on Pcengines Alix board

Il giorno Mon, 19 Nov 2007 08:54:38 -0500 Matthew Rubenstein [EMAIL PROTECTED] ha scritto: Other than the Alix board, what else is needed to make a working PC? You need a CF as main storage device (it is mounted ro on /). I also use an USB stick where I mount /var in rw mode. Obviously

Re: [asterisk-users] Asterisk on Pcengines Alix board

Date: Mon, 19 Nov 2007 10:39:31 -0600 From: Bob Pierce [EMAIL PROTECTED] Subject: Re: [asterisk-users] Asterisk on Pcengines Alix board To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain

Re: [asterisk-users] How to receive manager events from commands made by an AGI script?

That's because no event is being generated. I can do a quick patch for you and post it in mantis in order to accomplish that. But I am interested in know why you want to receive those events. I am in the middle of creating a new AGI application. As you probably know, you can launch AGI like this:

[asterisk-users] Music on Hold Problem w/ Transfers

I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. It does work when putting the call on hold. If I revert back to 1.2.23 using the same config and same music on hold files, it works.

Re: [asterisk-users] Registration problem: UA - SER - Asterisk

Stefano, It is not Asterisk, It is SER (dispatcher module ?). Why Asterisk is acting as Register ? make sense use openSER as Register/Proxy and Asterisk only Proxy and MG Regards, Giovanni 2007/11/19, Stefano Capitanio [EMAIL PROTECTED]: Hi, we a have a SER (OpenSER) in front of 2

Re: [asterisk-users] Music on Hold Problem w/ Transfers

On Tue, 2007-11-20 at 15:52 -0600, Lacy Moore wrote: I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. FWIW, I'm using 1.4.10 and music on hold for transfers is working fine... for

[asterisk-users] Asterisk-Users: Termination

I wanted to see if anyone has set up a large amount of out bound only voip channels? We run analog autodialers connected to analog to voip gateways (dialogic boards to audiocodes mp-124's) Bandwidth and routing is being provided by a bonded t-1 (3 meg) cisco 2600 with 2 t1 cards and a 16

Re: [asterisk-users] Music on Hold Problem w/ Transfers

On Nov 20, 2007 3:52 PM, Lacy Moore [EMAIL PROTECTED] wrote: I think I'm missing a change between 1.2 and 1.4. When using 1.4 (so far 1.4.9, 1.4.13, and 1.4.14), music on hold is not working for transfers or parked calls. It does work when putting the call on hold. If I revert back to

Re: [asterisk-users] Asterisk-Users: Termination

We are using only voip chanels with 400-500 channels. Although we are still in begining phase but i have not seen any problem as such. Thanks, Vivek On 11/20/07, Mark Adams [EMAIL PROTECTED] wrote: I wanted to see if anyone has set up a large amount of out bound only voip channels? We

[asterisk-users] Building an Asterisk 1.4 RPM

I'm a little confused. I'd like to build an RPM for Asterisk 1.4. Is it better to modify and use the spec file under redhat/asterisk.spec and run a 'make rpm', OR is it better to build a custom spec file from scratch and use 'rpmbuid -ba' specfile? How do people normally do it? The problem I

Re: [asterisk-users] Building an Asterisk 1.4 RPM

http://www.voip-info.org/tiki-index.php?page=Asterisk%20RPM Try this link. There is a lot of info and source rpms that you can rebuild. Jonn _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Douglas Garstang Sent: Tuesday, November 20, 2007 6:34 PM To:

[asterisk-users] quality after call transfer

Hi, We are using attended call transfer to transfer the call. In the direct call, the quality of the voice and dtmf are acceptable. After transfer, the quality becomes worst. Voice can't be heard clearly and dtmf wrong detection will occur sometime. I wonder call transfer will affect he

[asterisk-users] [1.4 - Record] How to tell if user did leave a msg?

Hello I didn't find the answer in the ATOF 2nd Ed: When using the Record() application, I need to know how it ended: Did the user leave a message, or did he hang up? If the latter, Asterisk stops right there, while I need to run some other commands before hanging up: exten =

Re: [asterisk-users] [1.4 - Record] How to tell if user did leave a msg?

page 511 use dialplan function STAT() -- On Nov 20, 2007 9:42 PM, Vincent wrote: Hello I didn't find the answer in the ATOF 2nd Ed: When using the Record() application, I need to know how it ended: Did the user leave a message, or did he hang up? If the latter, Asterisk stops right

[asterisk-users] Zaptel 1.4 spec file

Does anyone know where I can get an rpm spec file for zaptel 1.4.x? Thanks, Doug. Be a better sports nut! Let your teams follow you with Yahoo Mobile. Try it now.