Re: [asterisk-users] Templates

On Mon, Apr 11, 2011 at 09:37:08PM -0600, José Pablo Méndez Soto wrote: > Hi, > > Trying to create templates that allow higher compression of sip.conf, so for > example: > > [internal-number](!) > type=friend > secret=bigsecret > host=dynamic > context=internal > disallow=all > allow=ulaw > > [1

Re: [asterisk-users] Variable stripping/removing part of string

Weired result: exten => 0424449631,n,NoOp(${CALLERID(name)}) exten => 0424449631,n,NoOp(${${CUT(CALLERID(name),\(,1)}:0:-1}) -- Executing [0424449...@fax.inputinterior.se:4] NoOp("OOH323/Avaya2-248", "Martela (fax)") in new stack -- Executing [0424449...@fax.inputinterior.se:5] NoOp("OOH323/Avay

[asterisk-users] Templates

Hi, Trying to create templates that allow higher compression of sip.conf, so for example: [internal-number](!) type=friend secret=bigsecret host=dynamic context=internal disallow=all allow=ulaw [100](internal-extensions) mailbox=100@internal-extensions [101](internal-extensions) mailbox=101@inte

[asterisk-users] Poor call quality – line drop, chopping sound, like robotic voice, Both party could not hear caller voice

One of our client facing this issue, we have try to solve it but we're lack of asterisk knowledge. Anybody can help us? Isn't any problem with asterisk configuration or the problem come from PRI E1 itself? [Apr 11 15:32:48] VERBOSE[9231] chan_dahdi.c: -- Requested transfer capability: 0x00 - SPEEC

Re: [asterisk-users] Variable stripping/removing part of string

On Mon, 11 Apr 2011 12:58:39 +0200 wrote: > U were right, breaking it into two lines work. > > exten => 0424449631,n,NoOp(${CALLERID(name)}) > exten => 0424449631,n,Set(name=${CUT(CALLERID(name),\(,1)}) > exten => 0424449631,n,NoOp(${name:0:-1}) > -- Executing [0424449...@fax.inputinterior.se:4]

Re: [asterisk-users] Features.conf - Blind Transfer

Hi guys, I'm trying to get blind transfer to work and automatically transfer call to another number on key sequence press. Extensions.conf_snippet [from-pstn] exten => _0399377744,1,Set(__DYNAMIC_FEATURES=blindxfer) exten => _0399377744,n,Set(__GOTO_ON_BLINDXFR=to-pstn ^0388924326^1) exten =>

Re: [asterisk-users] Asterisk as a Condo door opener/intercom

Continuing top posting... The same argument could be made for any commercial solution. Why use Asterisk when we could throw $4,000 at our problem for a commercial solution? I'd like to have a solution that would have the features you suggest for $400. --Don On Behalf Of C F Sent: Monday, April

[asterisk-users] DAHDI-Linux 2.4.1.2 Released

The Asterisk Development Team announces the release of DAHDI-Linux 2.4.1.2. DAHDI-Linux 2.4.1.2 and DAHDI-Linux-Complete 2.4.1.2+2.4.1 are available for immediate download at: http://downloads.asterisk.org/pub/telephony/dahdi-linux http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete

Re: [asterisk-users] Voicemail to email issue

that is a sendmail issiue. Obviously asterisk is contacting 127.0.0.1 to try and deliver e-mail. Try help with sendmail folks, check that 127.0.0.1 is in the allowed to relay list or so.. On 11 April 2011 21:11, satish patel wrote: > Hi All, > > I have asterisk 1.8.3.2 and having issue with no

Re: [asterisk-users] [SOLVED] Voicemail to email issue

suck sendmail Solution: sudo apt-get remove sendmail sudo apt-get install ssmtp mailutils From: satish...@hotmail.com To: asterisk-users@lists.digium.com Date: Mon, 11 Apr 2011 20:11:27 + Subject: [asterisk-users] Voicemail to email issue Hi All, I have asterisk 1.8.3.2

[asterisk-users] Voicemail to email issue

Hi All, I have asterisk 1.8.3.2 and having issue with not getting VoiceMail email. I can send mail through command line using sendmail but not via asterisk. We have centralized zimbra email server. why its trying to send email to local 127.0.0.1 address? is there any other configuration i am m

Re: [asterisk-users] Asterisk MOH not working with Elastix asterisk1.6.2.18

Doesn't Elastix have it's own tool for MusicOnHold? Maybe check with that and see if that makes a difference. Thanks, --Warren Selby, dCAP On Apr 11, 2011, at 12:49 PM, virendra bhati wrote: > Yes , > > I show me the all configured MOH. But don't play the MOH. > > After 12 sec of silence C

Re: [asterisk-users] How to know the SIP status

On 4/11/2011 12:52 PM, virendra bhati wrote: > Hi , > > As we see the SIP shatus on CLI with *sip show status > > How we get the status with phpagi function ? > * > -- > - > Thanks and regards > > Virendra Bhati > +91-9172341457 > Googling is your friend: http://www.voip-info.org/wiki/view/

[asterisk-users] "Wait for leader" allows crosstalk between participants

Hello all, I was wondering if anybody found a solution for 0018418. It looks like any 1.8.x version is affect by bug #0018418. The meetme application allows crosstalk between participants when the "w" option (Wait for Leader) is enabled. Please let me know if anybody can help me with this. I have

[asterisk-users] How to know the SIP status

Hi , As we see the SIP shatus on CLI with *sip show status How we get the status with phpagi function ? * -- - Thanks and regards Virendra Bhati +91-9172341457 -- _ -- Bandwidth and Colocation Provided by http://www.api-d

Re: [asterisk-users] Asterisk MOH not working with Elastix asterisk1.6.2.18

Yes , I show me the all configured MOH. But don't play the MOH. After 12 sec of silence CLI give message Stop music on hold As I told you that default moh also not played :) On Mon, Apr 11, 2011 at 9:55 PM, Danny Nicholas wrote: >-- > > *From:* asterisk-us

Re: [asterisk-users] send voicemail to multiple emails

On 4/11/2011 12:47 PM, vip killa wrote: > Anyway, i figured out how to accomplish this using "externnotify"... > In app_voicemail.c, in the function "vm_execmain" i > commented out "run_externnotify(vmu->context, vmu->mailbox, NULL);" > Now "externnotify" is called by asterisk only when there is

Re: [asterisk-users] send voicemail to multiple emails

Anyway, i figured out how to accomplish this using "externnotify"... In app_voicemail.c, in the function "vm_execmain" i commented out "run_externnotify(vmu->context, vmu->mailbox, NULL);" Now "externnotify" is called by asterisk only when there is a new message and not when someone checks their vo

Re: [asterisk-users] send voicemail to multiple emails

I'm not confused about this... Everytime a voicemail is left, I need asterisk to run a script that will query a database, and according to those results perform various actions. These actions include calling a number and connecting it directly to voicemailmain and/or sending out multiple emails...

Re: [asterisk-users] Asterisk codec negotiation and canreinvite=no

On 11-04-11 10:26 AM, Effie Mouzeli wrote: This may lead to some buggy clients not to accept the call (with 488), but I've noticed some cases where a callee was behind NAT, an INVITE with one video codec would me forwarded properly to the callee, but another INVITE with 3 video codecs, would neve

Re: [asterisk-users] send voicemail to multiple emails

On 4/11/2011 12:30 PM, vip killa wrote: > Would be so much simpler if "mailcmd" acted just like "externnotify" > or "externnotify" only ran when a message was left but not when > someone checks their voicemail... > > > That's pretty much where you're at. What gets passed to STDIN is an >

Re: [asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?

You probably didn't read over my originally post carefully. In the dialplan A2billing.php script is called in the "X" extension. Then there is X,n,Hangup() so now "X" extension is dead. After that in "h" extension I have ANOTHER script running. However, the CLI output (which again I posted in my o

Re: [asterisk-users] send voicemail to multiple emails

Would be so much simpler if "mailcmd" acted just like "externnotify" or "externnotify" only ran when a message was left but not when someone checks their voicemail... > That's pretty much where you're at. What gets passed to STDIN is an > email, it's not set up for use by a script. Remember, what

Re: [asterisk-users] Unable to negotiate codec with iax

On 11-04-11 05:44 AM, Arjan Kroon | Mobillion wrote: We are using asterisk version 1.8.3.1 on the inbound server For the outbound server I used version 1.8.2.2. (If have tested with an inbound server with version 1.8.2.2 to the outbound server and that works fine.) Does anybody has an idea wha

Re: [asterisk-users] Asterisk as a Condo door opener/intercom

Search the lists. Some hints: Viking electronics makes a door box that connects to any analog line (IIRC e-20). They also make a DTMF keypad that integrates in series with any analog line. They might also make a door box with a DTMF keypad on it. Sandman makes a relay that will get energized when t

Re: [asterisk-users] Asterisk kernel CONFIG_HZ=1000

I have install ubuntu kernel 2.6.32-30-preempt to have 1000Hz timing. > Date: Mon, 11 Apr 2011 18:28:36 +0300 > From: tzafrir.co...@xorcom.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] Asterisk kernel CONFIG_HZ=1000 > > On Mon, Apr 11, 2011 at 02:44:54PM +, sa

Re: [asterisk-users] Occasional call from "asterisk"

On Mon, Apr 11, 2011 at 8:47 AM, Brian Henning wrote: > H. I do see this in the /var/log/asterisk/messages log: > > > > [Apr 5 00:05:36] NOTICE[9579] chan_dahdi.c: Got event 4 (Alarm)... > > [Apr 5 00:05:38] NOTICE[9579] chan_dahdi.c: Got event 5 (No more alarm)... > > [Apr 5 00:05:38] NOT

Re: [asterisk-users] Variable inheritance with dialplan command Originate

On 4/11/2011 5:15 AM, Naomi Rosenberg wrote: > > Hi, > > The reason I think Dial isn't appropriate is not to do with the database > call. Here's the wider context of the application I'm putting together: > > Punter calls in, leaves a message, gets a reference number, hangs up. System > then init

Re: [asterisk-users] Asterisk MOH not working with Elastix asterisk1.6.2.18

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati Sent: Monday, April 11, 2011 11:23 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Asterisk MOH not working with Elastix

Re: [asterisk-users] Asterisk MOH not working with Elastix asterisk 1.6.2.18

Hi All, I already try StartMusicOnHold() instead of MusicOnHold(); Even default asterisk MOH not playing. On Mon, Apr 11, 2011 at 9:17 PM, Warren Selby wrote: > Your last line in the dialplan should be StartMusicOnHold(), not just > MusicOnHold(). > > Thanks, > --Warren Selby, dCAP > > On A

Re: [asterisk-users] send voicemail to multiple emails

On 4/9/2011 11:56 PM, vip killa wrote: > I've already taken the steps you described...issue i ran into was > there is no variables passed to "mailcmd" only STDIN... as a result i > have to "extract" variables from STDIN... > > On Fri, Apr 8, 2011 at 5:09 PM, Warren Selby

Re: [asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?

h is hangup extension, and will be executed after hangup On Mon, Apr 11, 2011 at 6:36 PM, Bruce B wrote: > Thanks for the input but I am not sure if that answer my question of if > it's normal behaviour for AGI scrip to terminate after the "h" extension > rather than end of "x" extension even i

Re: [asterisk-users] Asterisk MOH not working with Elastix asterisk 1.6.2.18

Your last line in the dialplan should be StartMusicOnHold(), not just MusicOnHold(). Thanks, --Warren Selby, dCAP On Apr 11, 2011, at 6:24 AM, virendra bhati wrote: > I am using Elastix. Asterisk is used for PBX application in Elastix. I want > to make customize MOH. But not able to use MOH

Re: [asterisk-users] Asterisk kernel CONFIG_HZ=1000

On Mon, Apr 11, 2011 at 02:44:54PM +, satish patel wrote: > > Hey Guys! > > Just recently i come to know about this option CONFIG_HZ=1000 in kernel is > this important for asterisk application ? we have ubuntu with CONFIG_HZ=100 > should i think about this option ? > > root@shirley:/boo

Re: [asterisk-users] Asterisk kernel CONFIG_HZ=1000

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, April 11, 2011 9:58 AM To: asterisk-users Subject: Re: [asterisk-users] Asterisk kernel CONFIG_HZ=1000 Believe me i went through all google pages and

Re: [asterisk-users] Asterisk kernel CONFIG_HZ=1000

Believe me i went through all google pages and read every possible post. But i just wanted to know from asterisk point of view. -S From: da...@debsinc.com To: asterisk-users@lists.digium.com Date: Mon, 11 Apr 2011 09:50:36 -0500 Subject: Re: [asterisk-users] Asterisk kernel CONFIG_HZ=1000

Re: [asterisk-users] voicemail odbc "Length is ....."

Apologies you were correct, i had debug on... Sorry... On Mon, Apr 11, 2011 at 10:45 AM, Danny Nicholas wrote: >-- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *vip killa > *Sent:* Monday

Re: [asterisk-users] Asterisk kernel CONFIG_HZ=1000

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of satish patel Sent: Monday, April 11, 2011 9:45 AM To: asterisk-users Subject: [asterisk-users] Asterisk kernel CONFIG_HZ=1000 Hey Guys! Just recently i come to know about thi

Re: [asterisk-users] voicemail odbc "Length is ....."

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of vip killa Sent: Monday, April 11, 2011 9:44 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] voicemail odbc "Length is ." indeed

[asterisk-users] Asterisk kernel CONFIG_HZ=1000

Hey Guys! Just recently i come to know about this option CONFIG_HZ=1000 in kernel is this important for asterisk application ? we have ubuntu with CONFIG_HZ=100 should i think about this option ? root@shirley:/boot# cat config-2.6.32-30-server | grep CONFIG_HZ_ CONFIG_HZ_100=y # CONFIG_HZ_2

Re: [asterisk-users] voicemail odbc "Length is ....."

indeed but why in console and the info is so limited, it doesn't say which message or anything...strange On Mon, Apr 11, 2011 at 10:31 AM, Steven Howes wrote: > On 11 Apr 2011, at 15:28, vip killa wrote: > > I'm using voicemail ODBC with Asterisk 1.6.2.17.2. > > Why do I see "Length is 18

Re: [asterisk-users] voicemail odbc "Length is ....."

On 11 Apr 2011, at 15:28, vip killa wrote: > I'm using voicemail ODBC with Asterisk 1.6.2.17.2. > Why do I see "Length is 186545" or something similar but a different number > in Asterisk CLI everytime someone leaves a message? Because not all messages are the same length. I'd guess it's length

[asterisk-users] voicemail odbc "Length is ....."

I'm using voicemail ODBC with Asterisk 1.6.2.17.2. Why do I see "Length is 186545" or something similar but a different number in Asterisk CLI everytime someone leaves a message? -- _ -- Bandwidth and Colocation Provided by http://

[asterisk-users] Asterisk codec negotiation and canreinvite=no

Hi all, I realise that asterisk's codec negotiation has been discussed in the past multiple times. What I haven't been able to understand is how asterisk decides which video codecs to advertise to the other end when canreinvite=no in sip.conf and the initial caller doesn't support video. My

Re: [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]

In my kernel i found following options. its configured for 100Hz do you think if i set it to 1000Hz it will be more responsive ? root@shirley:/boot# cat config-2.6.32-30-server | grep CONFIG_HZ_ CONFIG_HZ_100=y # CONFIG_HZ_250 is not set # CONFIG_HZ_300 is not set # CONFIG_HZ_1000 is not set

Re: [asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?

Thanks for the input but I am not sure if that answer my question of if it's normal behaviour for AGI scrip to terminate after the "h" extension rather than end of "x" extension even if it was only run in "x" extension. Regards, On Mon, Apr 11, 2011 at 6:34 AM, Pezhman Lali wrote: > Dear > ther

[asterisk-users] Require dialplan

Hi , In vicidial dialer I need small Dialplan require. when i call from hardphone , in that has 1to9 no.s i want define the dipositions like when i press the 1 it will goes NotIntrest, press 2 for NotAvailable. How can i configure for this. -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business S

Re: [asterisk-users] Occasional call from "asterisk"

I wonder if you can test to see if this happens if you had an analogue phone set connected. And if it doesn't then I am wondering why Asterisk or Sangoma card is so sensitive and maybe the sensor can be set a bit higher so these calls don't end-up ringing like they don't if an analogue phone set wa

Re: [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]

On Mon, Apr 11, 2011 at 01:49:59PM +, satish patel wrote: > > > @ Tzafrir > > you mean say i shouldn't use "-server" kernel for asterisk ? I meant that a different kernel flavour may work better. Though I asked for feedback about that. -- Tzafrir Cohen icq#16849755

Re: [asterisk-users] Occasional call from "asterisk"

_ From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Brian Henning Sent: Monday, April 11, 2011 8:47 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Occasional call from "asterisk" B

[asterisk-users] Problem with E1 (ISDN) + DTMF

Hello, I have the scenario: Link E1 with ISDN running with asterisk 1.8.2.3, voicerlib 4.2.3.0, libpri 1.4.11.4 and dgvchannel 1.0.8. Asterisk begins a call, when the called party answer the call, it asks for the the called party to input some numbers (via DTMF) but when the numbers are presse

Re: [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]

@ Tzafrir you mean say i shouldn't use "-server" kernel for asterisk ? -Satish Date: Mon, 11 Apr 2011 07:45:01 -0400 From: stot...@asteriskhelpdesk.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199] On Sun, Apr

Re: [asterisk-users] Asterisk-Asterisk E1 connection

You need a E1/T1 crossover cable, which isn't straight through or like a network crossover cable. Search online for T1 crossover and you'll find the pinout. Remember one node needs to be the clock source (and only one node). Technically UTP isn't the right cable for E1/T1s, but if your distanc

Re: [asterisk-users] Asterisk-Asterisk E1 connection

For PRI coross over cable following is pin layout 1 <---> 4 2 <---> 5 > Date: Mon, 11 Apr 2011 10:43:51 -0300 > From: aco1...@gmail.com > To: asterisk-users@lists.digium.com > Subject: [asterisk-users] Asterisk-Asterisk E1 connection > > Dear, I have two Asterisk PBXs with an E1 interface/RJ-4

Re: [asterisk-users] Occasional call from "asterisk"

Bruce B said: We experience exact same thing on DAHDI with Sangoma USB FXO device on short circuited lines. Phantom calls are actually due to a short in the lines that happen occasionally. -Bruce Also, Warren Selby said: I've seen this on cases where a "phantom" call comes in on a DAHDI

[asterisk-users] Asterisk-Asterisk E1 connection

Dear, I have two Asterisk PBXs with an E1 interface/RJ-45 port in both boxes. I need to connect both PBXs with E1/R2 and UTP cable. What are the requirements to deploy the UTP cable ??? Straight-through or crossover ??? What are the pinouts in both peers ??? Thanks a lot, Alejandro -- _

Re: [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]

I don't understand what you guys talking about? You mean say there is a issue in ubuntu kernel to use asterisk? -- Sent from my iPhone On Apr 11, 2011, at 8:05 AM, Tzafrir Cohen wrote: On Mon, Apr 11, 2011 at 07:45:01AM -0400, Steve Totaro wrote: On Sun, Apr 10, 2011 at 9:12 AM, Tzafrir

Re: [asterisk-users] send voicemail to multiple emails

We are talking about "mailcmd" not "externnotify" I am aware of extennotify, problem is, it runs script when someone checks their voicemail, i need a script to run only when a voicemail is left On Mon, Apr 11, 2011 at 6:32 AM, Andrew Thomas wrote: > Not quite true. I use a PHP script to do

Re: [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]

On Mon, Apr 11, 2011 at 07:45:01AM -0400, Steve Totaro wrote: > On Sun, Apr 10, 2011 at 9:12 AM, Tzafrir Cohen > wrote: > > > Off-topic: > > > > On Fri, Apr 08, 2011 at 03:30:58PM +, satish patel wrote: > > > > [snip] > > > > > System: Linux/2.6.32-24-server built by ro

Re: [asterisk-users] Ubuntu "*-server" kernels [was: Re: IAX2/0.0.29.199]

On Sun, Apr 10, 2011 at 9:12 AM, Tzafrir Cohen wrote: > Off-topic: > > On Fri, Apr 08, 2011 at 03:30:58PM +, satish patel wrote: > > [snip] > > > System: Linux/2.6.32-24-server built by root on > x86_64 2011-03-22 18:38:19 UTC > > Ubuntu has a separate -server kernel var

[asterisk-users] Asterisk MOH not working with Elastix asterisk 1.6.2.18

I am using Elastix. Asterisk is used for PBX application in Elastix. I want to make customize MOH. But not able to use MOH. I make simple extension in asterisk conf file but no success :( But when I used Vanilla Asterisk then All things are working Below are the details of configuration files

Re: [asterisk-users] Variable stripping/removing part of string

U were right, breaking it into two lines work. exten => 0424449631,n,NoOp(${CALLERID(name)}) exten => 0424449631,n,Set(name=${CUT(CALLERID(name),\(,1)}) exten => 0424449631,n,NoOp(${name:0:-1}) -- Executing [0424449...@fax.inputinterior.se:4] NoOp("OOH323/Avaya2-150", "Martela (fax)") in new sta

Re: [asterisk-users] Is it the normal behaviore for AGI and DeadAGI to terminate AFTER the "h" extension?

Dear there is some problem. the true way for running php script, is using agi not system. second after 5 sec, a lot of channel variables were removed, it makes your program wrong. with some little experience you can add your script to a2billing, try it. best On Sat, Apr 9, 2011 at 7:22 PM, Bruce

Re: [asterisk-users] send voicemail to multiple emails

Not quite true. I use a PHP script to do my processing (called from voicemail.conf [externnotify = /usr/local/bin/vmnotify.php]). The main three lines are: $vm_context = $argv[1]; $extension = $argv[2]; $number_of_messages = $argv[3]; Self explanatory really. -Original Message- Fro

[asterisk-users] update CDR fields after Queue

Dears; I have been faced with a problem that I am not sure about how can I solve it... I my scenario there is a variable which will be ready just after the callee had hanged up and the caller, which coming throw a Queue. But the CDR fields are logged into DB just after the Queue application. so th

Re: [asterisk-users] Variable inheritance with dialplan command Originate

Hi, The reason I think Dial isn't appropriate is not to do with the database call. Here's the wider context of the application I'm putting together: Punter calls in, leaves a message, gets a reference number, hangs up. System then initiates call to a queue of on-call staff and when one answers

Re: [asterisk-users] Variable stripping/removing part of string

Hi Magnus, > exten => 0424449631,n,NoOp(${CUT(CALLERID(name),\(,1):0:-1}) > > But that gave me “Martela “ so my way of doing it is wrong. > Any that can tell me what I am doing wrong or have any better suggestion > howto do it? I think you are not able to do it in one step. Can you try something

Re: [asterisk-users] changing port 5060 to 5061

On 11 Apr 2011, at 10:03, darin iv wrote: > please send me the ways to change asterisk port from 5060 to 5061 i > need to configure it because we are already using 5060 port in router > then we cant use it again we have to configure other sip server so > please suggest me a way.

[asterisk-users] Fwd: changing port 5060 to 5061

-- Forwarded message -- From: darin iv Date: Mon, 11 Apr 2011 14:33:24 +0530 Subject: changing port 5060 to 5061 To: asterisk-users@lists.digium.com Dear Experts, please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using

[asterisk-users] changing port 5060 to 5061

Dear Experts, please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.. -- _

Re: [asterisk-users] Variable stripping/removing part of string

I dont know if my mail client will keep formatting as I se it, but for me it sure looks like one space. -- Executing [0424449...@fax.inputinterior.se:4] NoOp("OOH323/Avaya2-109", "Martela (fax)") in new stack xyz -

Re: [asterisk-users] send voicemail to multiple emails

On Friday 08 Apr 2011, vip killa wrote: > Is there a way for asterisk's voicemail to send an email (including > voicemail attachment) to multiple email addresses? It's probably easiest to set up a user on your mail server to receive the voicemail messages that are meant for multiple recipients, t

Re: [asterisk-users] Variable stripping/removing part of string

On Monday 11 April 2011 02:56:03 magnu...@inputinterior.se wrote: > It was a 1.8 but then we started to do a lot of development (ooh323) so > today it is Asterisk SVN-may-ooh323_ipv6_direct_rtp-r311741MS-/trunk. > Can hardly se that we have done any changes that would cause my > "problem". Are you

Re: [asterisk-users] Variable stripping/removing part of string

It was a 1.8 but then we started to do a lot of development (ooh323) so today it is Asterisk SVN-may-ooh323_ipv6_direct_rtp-r311741MS-/trunk. Can hardly se that we have done any changes that would cause my "problem". -Ursprungligt meddelande- From: Tilghman Lesher Sent: Monday, April 1

Re: [asterisk-users] Variable stripping/removing part of string

On Monday 11 April 2011 00:25:35 magnu...@inputinterior.se wrote: > Now i am lost. > exten => 0424449631,n,NoOp(${CALLERID(name)}) > exten => 0424449631,n,NoOp(${CUT(CALLERID(name),\(,2):0:-1}) > -- Executing [0424449...@fax.inputinterior.se:4] NoOp("OOH323/Avaya2-8", > "Martela (fax)") in new stac

Re: [asterisk-users] MWI not working on most ATAs in Asterisk 1.6.2.17

maill...@lightspeed.ca writes: > We've had several customers report since upgrading them to our new > Asterisk 1.6.2.17 server (from version 1.4), that their MWI no longer > works. No significant changes have been made to their SIP > configuration, nor to their ATA configuration. My testing of 1.