Re: [asterisk-users] Is it possible to use Stasis to control both legs of a Local channel created using ARI?

ext=1=mycallerid.1=mycallerid.2=ulaw=30; Have a good day! Dan From: asterisk-users On Behalf Of Joshua C. Colp Sent: Thursday, August 6, 2020 12:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to use Stasis to control both legs of a

Re: [asterisk-users] Is it possible to use Stasis to control both legs of a Local channel created using ARI?

On Thu, Aug 6, 2020 at 1:52 PM Dan Cropp wrote: > I understand how to control the first local channel, but an having trouble > getting the second local channel to enter stasis. > > > > I setup have the following extensions.conf to handle 1000 (basically had > it setup so if first stasis not

Re: [asterisk-users] Is it possible to have a single AMI originate ring multiple contacts?

Thank you Joshua. From: asterisk-users On Behalf Of Joshua C. Colp Sent: Wednesday, May 27, 2020 5:23 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to have a single AMI originate ring multiple contacts? On Wed, May 27, 2020 at 5

Re: [asterisk-users] Is it possible to have a single AMI originate ring multiple contacts?

On Wed, May 27, 2020 at 5:30 PM Dan Cropp wrote: > I have an endpoint with multiple phones registered as aor contacts. > > > > When I attempt to originate a call it will only ring one of the phones. > > > > Is it possible to ring multiple phones as a single endpoint. First phone > to answer

Re: [asterisk-users] Is it possible to record 2-4 party call audio in stereo quality as opposed to mono?

other participants (from ConfBridge). I will give the MixMonitor a try. Have a great day! Da -Original Message- From: asterisk-users On Behalf Of Antony Stone Sent: Friday, November 1, 2019 5:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk

Re: [asterisk-users] Is it possible to record 2-4 party call audio in stereo quality as opposed to mono?

On Friday 01 November 2019 at 22:29:28, Dan Cropp wrote: > We have a customer who wants us to record anywhere from 2-4 participants on > a call in stereo (as opposed to mono) quality audio. I'm assuming you mean you want to get one stereo recording for each participant, where the left channel

Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

Thanks George From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of George Joseph Sent: Thursday, January 04, 2018 8:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to have two

Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

Thank you George From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of George Joseph Sent: Thursday, January 04, 2018 8:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to have two

Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

for that... https://issues.asterisk.org/jira/browse/ASTERISK-27548 > > > *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] *On Behalf Of *George Joseph > *Sent:* Tuesday, December 19, 2017 7:57 AM > > *To:* Asterisk Users Mailing L

Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

George Joseph Sent: Tuesday, December 19, 2017 7:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that

Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

e-ordered. [1] https://issues.asterisk.org/jira/browse/ASTERISK-27491 > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- > boun...@lists.digium.com] *On Behalf Of *George Joseph > *Sent:* Thursday, December 14, 2017 10:59 AM > *To:*

Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

On Mon, Dec 18, 2017, at 12:04 PM, Dan Cropp wrote: > Thanks George > > I originally didn’t have the 1002@ for the identify. Changed that when > things were not working. I changed it back. > > Unfortunately, the system I am connecting with doesn’t seem to support > the line support. Looking

Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

] On Behalf Of George Joseph Sent: Thursday, December 14, 2017 10:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk

Re: [asterisk-users] Is it possible to have two endpoints to the same IP address where one uses IP based authentication and the other requires asterisk to register to that system?

On Wed, Dec 13, 2017 at 10:51 AM, Dan Cropp wrote: > Currently using PJSIP. First, they want me to get this working with the > existing PJSIP configuration, but then setup a second box using chan_sip > performing similar work. > > > > For PJSIP… > > I currently have an

Re: [asterisk-users] Is it possible that variables returned from AGI take a moment to "stick"?

I don't think so any such method to return variable from AGI. But simple solution is set variable in AGI and then you can get back after AGI call in dialplan and these variable will be available until call finished. --- Virendra Bhati +91-9718500594 +91-9250078532 Sr. Asterisk Developer

Re: [asterisk-users] Is it possible to set a timeout when querying a calendar ?

2016-06-21 12:18 GMT+02:00 Olivier : > Hello, > > Asterisk offers several ways to query an external calendar. > I'm wondering what could happen if an external calendar was down or very > slow to respond. > > Looking at wiki.asterisk.org, I didn't find any timeout parameter in

Re: [asterisk-users] Is it possible to have two trunks between two Asterisk boxes ?

Olivier wrote: So basically, I then must use an other header (than From header) to pass Caller IDs between the two boxes, no ? Yes. Which header is then recommanded ? P-Asserted-Identity ? Yes. That or RPID. Both are supported. Is this commonly supported and configurable by non-Asterisk

Re: [asterisk-users] Is it possible to have two trunks between two Asterisk boxes ?

2016-04-05 17:12 GMT+02:00 Joshua Colp : > Olivier wrote: > >> Hello, >> >> For lab testing, I'm trying to build two differents PJSIP trunks between >> two Asterisk 13.8.0enabled boxes. >> I thought I could set up both trunks like this: >> Box A/port 5060 <-- Trunk1 ->

Re: [asterisk-users] Is it possible to have two trunks between two Asterisk boxes ?

Olivier wrote: Hello, For lab testing, I'm trying to build two differents PJSIP trunks between two Asterisk 13.8.0enabled boxes. I thought I could set up both trunks like this: Box A/port 5060 <-- Trunk1 -> Box B/port 5060 Box A/port 5062 <-- Trunk2 -> Box B/port 5062 and

Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

Thank you Niklas That solved my problem. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Niklas Larsson Sent: Friday, August 28, 2015 1:55 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Is it possible to perform

Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

-boun...@lists.digium.com] On Behalf Of Dan Cropp Sent: Friday, August 28, 2015 8:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? Thank you Niklas

Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? To add a header to the call leg that goes to the agent, try using a local channel to activate dialplan on the outbound call: Register Local/number@agent

Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

To add a header to the call leg that goes to the agent, try using a local channel to activate dialplan on the outbound call: Register Local/number@agent in the queue on behalf of the agent (replace number with the agent's extension number) In dialplan [agent], wild card match the number, add the

Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

*To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? Local channels: http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk

Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

: asterisk-users-boun...@lists.digium.com] *On Behalf Of *Scott Griepentrog *Sent:* Thursday, August 27, 2015 1:57 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part

Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet?

Of Scott Griepentrog Sent: Thursday, August 27, 2015 4:28 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to perform PJSIP Add Header prior to calling Queue and have it part of the INVITE packet? Local channels: http

Re: [asterisk-users] Is it possible to set asterisk's VoIP authentication to be based on EAP-SIM auth of freeradius?

Hi Shishir, thank you for your guide, hope this could help me a lot. Rafa -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs:

Re: [asterisk-users] Is it possible to set asterisk's VoIP authentication to be based on EAP-SIM auth of freeradius?

Hi Shishir, thanks your your response, would you help me about how to set up in the sip proxy server? actually, i'm beginner on asterisk. thank you. Hi Jaya, it would be nice for me if i can assist you, but i don't know to much about asterisk. i'm sorry On Fri, Aug 8, 2014 at 3:05 AM,

Re: [asterisk-users] Is it possible to set asterisk's VoIP authentication to be based on EAP-SIM auth of freeradius?

Of rafa alfurqan Sent: Friday, August 08, 2014 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Is it possible to set asterisk's VoIP authentication to be based on EAP-SIM auth of freeradius? Hi Shishir, thanks your your response, would you help me about

Re: [asterisk-users] Is it possible to set asterisk's VoIP authentication to be based on EAP-SIM auth of freeradius?

You can use sip proxy servers on top of asterisk server to have a authentication from freeradius, at this point I don’t think asterisk supports what you are looking for. Try this http://www.opensips.org/Documentation/Tutorials-Radius From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] is it possible to compile chan_h323 with 11.5.0?

ok, changed to ast_channel_writeformat ast_channel_readformat at least, got it compiled :-D 22.08.2013 13:24, Dmitry Melekhov пишет: Hello! Tried to compile, but : [CC] chan_h323.c - chan_h323.o chan_h323.c: In function '__oh323_update_info': chan_h323.c:349: error: dereferencing

Re: [asterisk-users] Is it possible to provision lock Aastra phones?

On 07/06/2013 03:35 PM, Bruce B wrote: Thanks Patrick. Do the encrypted config files safe guard against hard resets such as Web Recovery mode - aka holding down 1 # sign at startup? My main purpose is to lock the sets due to contract terms so I'd rather not see user steal the phone and break

Re: [asterisk-users] Is it possible to provision lock Aastra phones?

On 07/06/2013 08:15 AM, Bruce B wrote: Hi everyone; Is it possible to provision lock Aastra phones to provider so that no soft or hard reset can unlock them? Iirc you can use encrypted configs using an app called anacrypt and lock them down. The admin guide (3.2.2) has more details in

Re: [asterisk-users] Is it possible to provision lock Aastra phones?

Thanks Patrick. Do the encrypted config files safe guard against hard resets such as Web Recovery mode - aka holding down 1 # sign at startup? My main purpose is to lock the sets due to contract terms so I'd rather not see user steal the phone and break contract without payment. Regards On

Re: [asterisk-users] Partial authentication possible?

Consider using a sip proxy server such as OpenSIPS or Kamailio. Regards, Ali Pey On Mon, Dec 10, 2012 at 12:59 PM, John Gilbert jo...@motorolasolutions.comwrote: I have a non-standard SIP client that I am trying to integrate with an Asterisk 10 server. This client requires that it

Re: [asterisk-users] If would possible use a custom function in Asterisk Dialplan

Hey Longst, I'd recommend having a look into the LUA support Asterisk offers for its dialplans or AGI. These are the only realistic ways to add functions, unless you want to write your own C module and compile it in. Adhearsion is an option as well, if you are proficient with ruby. Cheers,

Re: [asterisk-users] If would possible use a custom function in Asterisk Dialplan

You could do it as a function if you are C literate. The simpler way would be to do it as an AGI where you passed the ${EXTEN} value to the AGI and had the AGI pass the modified number back as a dialplan variable. -Original Message- From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Is it possible call land into extensions.ael configuration file not in extensions.conf

Hi, Create a context in AEL, or LUA and change the context=ael-context or context=lua-context in sip.conf [default] section or for each sip user decalred who needs to start call in context defined in AEL/LUA? Regards, Gohar From: asterisk-users-boun...@lists.digium.com

Re: [asterisk-users] Is it possible call land into extensions.ael configuration file not in extensions.conf

Hi Gohar, As per you suggestion I make context into AEL file and working file. But I do little bit RD on that case I make same context into both files(.conf and .ael) and asterisk read 1st .conf files extension. It means if we make anythings into AEL files then asterisk 1st check into .conf file

Re: [asterisk-users] Is it possible call land into extensions.ael configuration file not in extensions.conf

Sorry for the top post, this is from my phone. Asterisk parses all of the config files (.conf, .ael and .lua, assuming you have the appropriate modules loaded) at the time you load asterisk or reload the dialplan (dialplan reload). It does not read the files each time a new call is started.

Re: [asterisk-users] Is it possible to dial an automated message when Asterisk receives an email?

You could use a procmail recipe to create a call file and then move it to the /var/spool/asterisk/outgoing directory. Below is a untested example .procmailrc: :0: * ^to.trig...@example.com | /usr/local/bin/callout.sh where callout.sh would look like this perhaps: !/bin/bash sleep 5

Re: [asterisk-users] Is it possible to dial an automated message when Asterisk receives an email?

John, Thanks for your reply. I will test this script. Once again, thank you! On Sat, Apr 2, 2011 at 3:39 PM, John Kiniston j...@meatkite.com wrote: You could use a procmail recipe to create a call file and then move it to the /var/spool/asterisk/outgoing directory. Below is a untested

Re: [asterisk-users] Is it possible to dial an automated message when Asterisk receives an email?

On Sat, 2 Apr 2011, Rafael Bermúdez wrote: I have a server that sends a preformatted email when an event occur. What I need is that when Asterisk receives this email automatically dial a pre-recorded message. It doesn't have to dial ride away, maybe a scheduled cron job will be enough.

Re: [asterisk-users] Is it possible to dial an automated message when Asterisk receives an email?

Un-top-posting... On Sat, Apr 2, 2011 at 3:39 PM, John Kiniston j...@meatkite.com wrote: You could use a procmail recipe to create a call file and then move it to the /var/spool/asterisk/outgoing directory. Below is a untested example .procmailrc: :0: *

Re: [asterisk-users] Is it possible to dial an automated message when Asterisk receives an email?

Steve, Thanks for your advice! You have an interesting point of view. I shall discuss this with my office partners on Monday. Thank you both. PS: In case you wonder, I'm from Argentina, hence my native language is Spanish On Sat, Apr 2, 2011 at 3:57 PM, Steve Edwards

Re: [asterisk-users] Is it possible to keep both call legs live after Dial()

On 07/09/10 04:15, C F wrote: Dial with M option I don't see how executing a macro BEFORE connecting to the called channel will help with my issue. Am I missing something? Thanks, Barry -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Is it possible to keep both call legs live after Dial()

On Tuesday 07 September 2010 04:29:54 Barry O'Donovan wrote: On 07/09/10 04:15, C F wrote: Dial with M option I don't see how executing a macro BEFORE connecting to the called channel will help with my issue. Am I missing something? You're missing that the macro is executed on the CALLED

Re: [asterisk-users] Is it possible to keep both call legs live after Dial()

From the way you posted your questions I understood that you want to give the option for the person being called to enter DTMF then hangup and keep the calling person alive (continue in dial plan) the M option will do that for you. http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial On Tue, Sep

Re: [asterisk-users] Is it possible to keep both call legs live after Dial()

On Mon, Sep 6, 2010 at 10:02 AM, Barry O'Donovan barry+asterisk-us...@opensolutions.ie wrote: Now, ideally, I would be able to act on a 'decision' from a DTMF sequence from the agent's handset. I don't think this is possible unfortunately. Please correct me if I'm wrong. DYNAMIC_FEATURES

Re: [asterisk-users] Is it possible to keep both call legs live after Dial()

Dial with M option On Mon, Sep 6, 2010 at 10:02 AM, Barry O'Donovan barry+asterisk-us...@opensolutions.ie wrote: Hi folks, After a fairly extensive Google trawl, I don't think the following is possible but would appreciate confirmation from anyone else who has tried something similar. I

Re: [asterisk-users] is it possible to connect Digium TE420 and Cisco card?

just FYI, to complete the topic. The problem was caused by failed PVDM module in Cisco server. Hello, I'm having problem connecting my Asterisk 1.4.29.1 with Digium TE420 to providers Cisco 2800 with VWIC-1MFT-E1 card. the same card runs fine with another E1 provider. TE420 led's

Re: [asterisk-users] GotoIfTime problem - possible bug

On Monday 23 November 2009 12:11:02 pm Nic Colledge wrote: I'm currently doing some testing against asterisk 1.6.1.7-rc2 (keep meaning to upgrade) and am having a problem with the GotoIfTime dial plan function. The asterisk book says that day of week field can include the ampersand () to

Re: [asterisk-users] GotoIfTime problem - possible bug

On Mon, Nov 23, 2009 at 1:11 PM, Nic Colledge n...@njcolledge.net wrote: I’m currently doing some testing against asterisk 1.6.1.7-rc2 (keep meaning to upgrade) and am having a problem with the GotoIfTime dial plan function. The asterisk book says that day of week field can include the

Re: [asterisk-users] GotoIfTime problem - possible bug

Discussion Subject: Re: [asterisk-users] GotoIfTime problem - possible bug On Monday 23 November 2009 12:11:02 pm Nic Colledge wrote: I'm currently doing some testing against asterisk 1.6.1.7-rc2 (keep meaning to upgrade) and am having a problem with the GotoIfTime dial plan function. The asterisk

Re: [asterisk-users] Is it possible to do this? (forward a call w/3-way calling)?

It may be possible. There is a flash application in Asterisk. Tom -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Justin Piszcz Sent: Saturday, June 13, 2009 9:59 AM To: asterisk-users@lists.digium.com

Re: [asterisk-users] Is it possible to get full callin number from E1?

thanks all i found the telco only send me the normal number 87654321 i just want to start a fax service and people can direct dial some extend num like 87654321...but it never send to me ... so the only thing i can do is to provide a ivr and let the people enter the extend num

Re: [asterisk-users] Is it possible to get full callin number from E1?

ssmax, Use CALLERID(num) to get the number that was dialed. Jimmy -Original Message- From: ss...@126.com Sent: Thu, 12 Mar 2009 20:40:59 +0800 To: asterisk-users@lists.digium.com Subject: [asterisk-users] Is it possible to get full callin number from E1? Hi all i have

Re: [asterisk-users] Is it possible to get full callin number fromE1?

: [asterisk-users] Is it possible to get full callin number fromE1? ssmax, Use CALLERID(num) to get the number that was dialed. Jimmy -Original Message- From: ss...@126.com Sent: Thu, 12 Mar 2009 20:40:59 +0800 To: asterisk-users@lists.digium.com Subject: [asterisk-users

Re: [asterisk-users] Is it possible to get full callin number fromE1?

Sorry, I indicated the wrong variable. You can always ask your provider what is sent. -Original Message- From: ss...@126.com Sent: Thu, 12 Mar 2009 22:11:32 +0800 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Is it possible to get full callin number fromE1? hi

Re: [asterisk-users] Is it possible to get full callin number fromE1?

...@126.com Sent: Thu, 12 Mar 2009 22:11:32 +0800 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Is it possible to get full callin number fromE1? hi,Jimmy Godbout when +86 136 make a call to - +86 020 87654321 - asterisk the CALLERID(num

Re: [asterisk-users] Is it possible to get full callin number fromE1?

Subject: Re: [asterisk-users] Is it possible to get full callin number fromE1? hi,Jimmy Godbout when +86 136 make a call to - +86 020 87654321 - asterisk the CALLERID(num) will show the caller number +86136 the ${EXTEN} is the dialed number 87654321 i will try

Re: [asterisk-users] Is it possible to get full callin number fromE1?

Steve wrote: Speaking from T1 PRI and E1 PRI in West Africa, you tell the telco how many digits to send. Often times, at least in my experience, if not specified, they will only send the last four providing there are no conflicts. They should be able to send however many digits you require,

Re: [asterisk-users] Is it possible to get full callin number from E1?

ssmax wrote: Hi all i have just set up a asterisk in china, using DE410P and one E1 line and get a phone number like: +86 020 87654321 from my sp when somebody dial +86 020 87654321 , the asterisk will get the call in number by ${EXTEN} variable, but it can only get 87654321, no area

Re: [asterisk-users] Is it possible to get full callin number from E1?

Jimmy Godbout wrote: ssmax, Use CALLERID(num) to get the number that was dialed. CALLERID(num) is the calling number, not the called number klaus ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing

Re: [asterisk-users] Is it possible to get full callin number from E1?

2009/3/12 ssmax ss...@126.com Hi all i have just set up a asterisk in china, using DE410P and one E1 line and get a phone number like: +86 020 87654321 from my sp when somebody dial +86 020 87654321 , the asterisk will get the call in number by ${EXTEN} variable, but it can only

Re: [asterisk-users] is it possible to store vmsecrets outside of users.conf?

Klaus Darilion schrieb: Currently I provision user account in users.conf. But I do not like that VoiceMail writes to users.conf when the voicemail password is changed. Is there a possibility to store the vmsecret in another place? (another file or DB)? Yes, if you were using Realtime to

Re: [asterisk-users] is it possible to deactivate RTCP?

So, you don't want any media? No audio, video, just sip packets? If you just want a sip router with no media look into SER. Klaus Darilion wrote: Hi! Is it possible to deactivate RTCP? (I am using 1.6) thanks klaus ___ -- Bandwidth and

Re: [asterisk-users] is it possible to deactivate RTCP?

I think he wants to leave RTP turned on, but turn off RTCP statistics collection and offers. Sorry I don't have an answer for the actual question, though. Seems reasonable, though perhaps selectable on a per-connection basis. Is RTCP crashing your remote end? JT On Nov 7, 2008, at

Re: [asterisk-users] is it possible? 1 VOIP Provider Multiple registrations to multiple inbound contexts

Not true. When you register the /1234 on the end of the line sends it to that extension in the context you specified in the peer entry with context=. Thanks for your help.. Can you please re-explain maybe another way? I'm having trouble understanding what you mean by: context you specified

Re: [asterisk-users] is it possible? 1 VOIP Provider Multiple registrations to multiple inbound contexts

On Fri, Jun 27, 2008 at 2:20 AM, Steve Gladden [EMAIL PROTECTED] wrote: In other words how to match a registration to a peer or inbound context other that the single defined default. I've also been told back in the asterisk 1.2 days that it was not possible. Not true. When you register the

Re: [asterisk-users] is this possible..

Yeah, Asterisk I think would be more than capable of doing that. It'll need some work to glue it all together. A lot of this would be written as an AGI script, and PHP or so for the webpage part of it. Sounds fun! blackwater dev wrote: We currently have an application used by the trucking

Re: [asterisk-users] is this possible..

We currently have an application used by the trucking industry to find freight to move. Now, the trucker does a search around Boston (for example) and gets 100 loads returned. They start at the first and call the company who has the freight, the company may say, sorry, someone just booked that

Re: [asterisk-users] is this possible..

blackwater dev wrote: I'm head of RD for a dot com company and we are looking to create a prototype using asterisk. Basically we people who visit our site and search for goods listed by other people. Once something is found, a phone number is listed in the results and person A calls

Re: [asterisk-users] is this possible..

I really see this is useless since we alreadu got pricegrabbers buy.com and froogle they all list the itme in stock on the site there is really no need for a $30k a year operator to read it for the person. just my $0.02 On 3/6/08, blackwater dev [EMAIL PROTECTED] wrote: I'm head of RD for a dot

Re: [asterisk-users] is this possible..

I think he's talking about an automated system. It's definitely possible with asterisk, whether or not it's a good idea. I really see this is useless since we alreadu got pricegrabbers buy.com and froogle they all list the itme in stock on the site there is really no need for a $30k a year

Re: [asterisk-users] is this possible..

It's certainly possible, and I would be interested in helping you get it going. --Don Don Kelly PCF Corp Real Support for your Virtual Office TM 651 842-1000 888 Don Kell(y) 651 842-1001 fax _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of blackwater dev Sent:

Re: [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?

I use 4960 and 4638 gw but it's applicable for any patton gw ( analogue or isdn ) since it's always the same way of configuring - define ports - define interface - define services my config: 1 asterisk 1 patton ( let say 4638 ) the patton gw registers itself has a asterisk sip peer, inside the

Re: [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?

2008/1/10, Jean-Louis curty [EMAIL PROTECTED]: I use 4960 and 4638 gw but it's applicable for any patton gw ( analogue or isdn ) since it's always the same way of configuring - define ports - define interface - define services my config: 1 asterisk 1 patton ( let say 4638 ) the patton

Re: [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?

exactly isdn patton - eth/lan sip asterisk jl 2008/1/10, Olivier [EMAIL PROTECTED]: 2008/1/10, Jean-Louis curty [EMAIL PROTECTED]: I use 4960 and 4638 gw but it's applicable for any patton gw ( analogue or isdn ) since it's always the same way of configuring - define ports - define

Re: [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?

2008/1/10, Jean-Louis curty [EMAIL PROTECTED]: exactly isdn patton - eth/lan sip asterisk so why is misdn installed for ? it seems you don't have any ISDN card inside you Asterisk server. jl 2008/1/10, Olivier [EMAIL PROTECTED]: 2008/1/10, Jean-Louis curty [EMAIL PROTECTED]:

Re: [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?

ok sorry for the confusion created, I mean isdn network , in other word tdm, so tdm link connected to patton, patton connected in the lan via ethernet speaking sip, jl 2008/1/10, Olivier [EMAIL PROTECTED]: 2008/1/10, Jean-Louis curty [EMAIL PROTECTED]: exactly isdn patton - eth/lan sip

Re: [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?

So, I think in this case (Ethernet link), standard spandsp doesn't help as it needs a TDM board. But, as spandsp has recently gained T.38 support, it could help to build email2fax or fax2email but I have no experience myself in it. I would be very curious to know. Regards 2008/1/10, Jean-Louis

Re: [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?

Olivier wrote: So, I think in this case (Ethernet link), standard spandsp doesn't help as it needs a TDM board. Nope not the case at all. I have been doing fax--ATA--lan--Asterisk-email for quite some time without ANY zaptel interfaces. Zaptel creats the pseudo interface and that does the

Re: [asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?

2008/1/8, Jean-Louis curty [EMAIL PROTECTED]: Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, which patton product do you use ? how

Re: [asterisk-users] Is it possible to register without sending the password

Dear Philipp; Thanks for your kindly help. The log was for the sip endpoint registeration as following: Really destroying SIP dialog '[EMAIL PROTECTED]' Method: REGISTER And this message was coming peridically (maybe every time the endpoint trying to register). The endpoint was the Firefly

Re: [asterisk-users] Is it possible to register without sending the password

Dear Philipp; How can I add the verbose and debug to the consol entry in the logger.conf to be able to take logging about the attempt of registeration for the sip end point? Regards Bilal If secret enabled, then some endpoints can not register (maybe due to compatibility in reading the

Re: [asterisk-users] Is it possible to register without sending the password

bilal ghayyad wrote: How can I add the verbose and debug to the consol entry in the logger.conf to be able to take logging about the attempt of registeration for the sip end point? console = notice,warning,error,debug,verbose as explained in /etc/asterisk/logger.conf Regards, Philipp

Re: [asterisk-users] Is it possible to register without sending the password

Dear Philipp; Kindly find the part of the configuration as below: [general] allow=all disallow is comment by ( ; ). [bilal_sip] type=friend context=internal host=dynamic canreinvite=no dtmfmode=rfc2833 So where is the problem? The endpoint does not register and nothing appear on trace level

Re: [asterisk-users] Is it possible to register without sending the password?

bilal ghayyad wrote: If secret enabled, then some endpoints can not register (maybe due to compatibility in reading the negotiation packets), so what is the solution? I'm sure they can. Maybe you could tell the list which endpoints don't work? Also in SIP registration: why I do not see the

Re: [asterisk-users] Is it possible to Voicemail menus (not just audiofiles) ?

Olivier, You have two options. 1) Change the source code. 2) Pay a coder to give you the options. Also this mat be the lack of sleep talking but from what I remember there was talk about this before. Search the archives. Dovid - Original Message - From: Olivier To: Asterisk

Re: [asterisk-users] is it possible to use Asterisk voicemail as anouncement system only?

On 12/15/06, Michael Hamann [EMAIL PROTECTED] wrote: Hello, we are using asterisk in combination with the voicemail system. I´m just wondering if it is possible to switch the voicemail to an I am on holiday mode. This means that the unavailable message is played to the caller but no

Re: [asterisk-users] is it possible to use Asterisk voicemail as anouncement system only?

we are using asterisk in combination with the voicemail system. I´m just wondering if it is possible to switch the voicemail to an I am on holiday mode. Just use Play(recordedmsg) instead of voicemail ___ --Bandwidth and Colocation provided by

Re: [asterisk-users] Is this possible?

Dont Use Call Progress Instead Use The M Option In App Dial That Asks The User To Press A Button To Accept The Call On 11/21/06, shadowym [EMAIL PROTECTED] wrote: Anyone tried this, I put in an Asterisk/FreePBX phone system to replace one of those el cheapo Bizfon analog key systems. The

Re: [asterisk-users] Is it possible have multiple ip numbers for an extension?

On Mon, 2006-11-06 at 21:31 +0800, William Kenworthy wrote: Is it possible have multiple concurrent ip numbers for a single extension? How? I am using a laptop that I move around various local and remote networks so the IP numbers it uses varies. As I am on extension '205', I want be

Re: [Asterisk-Users] Is it possible to delete global variables

Tzafrir Cohen schrieb: On Tue, May 16, 2006 at 01:09:41AM +0200, Bastian Schern wrote: Hello, is it possible to delete global variables during runtime? Is setting the variable to an empty value good enough? How do you use it? I will use it to check if a Caller has already a open

Re: [Asterisk-Users] Is it possible to delete global variables

On Tue, May 16, 2006 at 01:09:41AM +0200, Bastian Schern wrote: Hello, is it possible to delete global variables during runtime? Is setting the variable to an empty value good enough? How do you use it? -- Tzafrir ___ --Bandwidth and Colocation

Re: [Asterisk-Users] Is it possible to turn off password for transfers on FOP

It's possible that by setting show_security_code=0 in op_style.cfg and setting security_code= in op_server.cfg that A) default password is blank B) you have nowhere to change it shrugmight work/shrug Moj Chuck Bunn wrote: Hi, Is it possible to turn off the request for a security code when

Re: [Asterisk-Users] Is it possible to reinvite twice?

Gabe, If my setup goes: Phone = asterisk = asterisk = PSTN termination provider Can I define canreinvite on both asterisk boxes so the phone call will go directly to the PSTN provider? Yes, you can reinvite multiple times. The media path will collapse as much as possible. It works reliably,

Re: [Asterisk-Users] Is it possible ?

Hi , I think i understand what you mean by your mail.I have done the same thing. You must download following modules from and asterisk site e.g. www.digium.com 1.asterisk-1.0.3.tar 2.libpri-1.0.3.tar 3.zaptel-1.0.3.tar Then there is a process youneed to follow which you will find in th read me

Re: [Asterisk-Users] Is it possible ?

thanks amna i have done it .. On 2/1/06, amna saleem [EMAIL PROTECTED] wrote: Hi , I think i understand what you mean by your mail.I have done the same thing. You must download following modules from and asterisk site e.g. www.digium.com 1.asterisk-1.0.3.tar 2.libpri-1.0.3.tar 3.zaptel-1.0.3.tar

  1   2   >