Re: [asterisk-users] IAX2/SIP gateway for Belgium and western Europe

On 19/01/07, Jan Dewerchin [EMAIL PROTECTED] wrote: Dear all, I'm not sure if this is the correct place to put it, but can I announce you the possibility of using a new, lost-cost trunk for Belgium and western Europe ? Maybe it's a shameless commercial plug, but have if you don't know it

Re: [asterisk-users] NAT solutions

On Thu, 18 Jan 2007, Voip Asterisk wrote: I know that NAT is something no one really likes to talk about, but does anyone know how work with it elegantly? There are many providers which deal with it on a daily basis in fact they cater to it, is this possible to do with asterisk or does it

[asterisk-users] mysterious SIP packets to Cogent

In my SOHO setting even when nobody is using the phone my firewall drops outgoing packets from the asterisk box to a Cogent server, din't find naything through Google about it: (out: eth0 xxx.xxx.xxx.xxx.:2129 - 66.250.40.33:24441 UDP len:193 ttl:64). Anyone know what this traffic is supposed

[asterisk-users] Voice Recognition

Hi all, Does anyone know if Asterisk or any available 3rd party add-on for it support voice recognition (not speech recognition) - task of recognizing people from their voices? Thanks, Alex ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [asterisk-users] Voice Recognition

My voice is my passport; verify me. ;) I don't think you'll get reliable results with 8khz sample rates. The highest frequency wave you can achieve is a 4khz square wave. Anyway, i don't think if such software exists ;) Julian J. M. On 1/19/07, Asterisk [EMAIL PROTECTED] wrote: Hi all,

Re: [asterisk-users] How to limit IAX calls

Take a look on: Dialplan applications: GetGroupMatchCount([EMAIL PROTECTED]) SetGroup([EMAIL PROTECTED]) Using this two applications you can deploy a max calls control inside your dialplan! check this too: http://www.voip-info.org/wiki/view/Asterisk+cmd+SetGroup Hope it helps On 1/19/07,

Re: [asterisk-users] NAT solutions

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Gordon Henderson wrote: If you only have one * box behind the NAT gateway then I don't really see a big issue with it to be honest. Port-forward on the firewall/router device (5060 and 1 through 2) to the * device, and use STUN on the

[asterisk-users] meetme ${DATETIME} variable update

Hi i am experiencing this problem: MEETME_RECORDINGFILE=/data/asterisk_data/_${DATETIME}_CONFERENCE exten = ,1,MeetMe(666|1Arxq) exten = 9998,1,MeetMe(666|1Axq) exten = 9997,1,MeetMe(666|1xq) I make a conference between 3 person dialing A dials B dials 9998 C dials 9997 all works

[asterisk-users] pickup call out of menu

Is it possible to pickup a caller, who is in the menus somewhere, for instance he may be lost in the telemarketer torture script? Just like it is possible to pick up a call on a ringing phone. Leif ___ --Bandwidth and Colocation provided by

[asterisk-users] direct transfer in features

I have some siemens wireless ip-phones. There is no problem entering ** which I have configured in features.conf to be transfer. But then it is difficult to enter the extension, because one have to wait the right amount of time before entering the extension. Because we only have few

Re: [asterisk-users] Sip Phone CID

Rob Schall wrote: This might sound like an odd question but here it is anyways... We currently have Polycom 501 phones. We have Asterisk with Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone dials another, the receiving end does in fact see the callers ID. But...

RE: [asterisk-users] How to limit IAX calls

A demonstration: exten = _X.,1,Set(GROUP()=${CALLERID(num)) exten = _X.,n,Set(CDR(AccountCode)=${CALLERID(num)) exten = _X.,n,GotoIf($[${GROUP_COUNT(${CALLERID(num))} 2]?103) exten = _X.,n,Macro(trunk,${EXTEN},residential) exten = _X.,n,Hangup exten =

RE: [asterisk-users] Voice Recognition

Hi Alex, I've spoken to some commercial (read 'large company') RD people who were messing around with telephony based voice recognitionnot great results and project was abandoned (basically the confirmation threshold was going to have to be set so low it wasn't worth it). Regards,

[asterisk-users] DND - message

Thanks very much for your response Andrew... Andrew Joakimsen wrote: Well now that I look into it, if you disable the call waiting the response is 486 busy here. If you use the DND it is the same response, so there's no way to do phone-side DND and correctly report the voicemail state.

[asterisk-users] Re: meetme ${DATETIME} variable update

On 1/19/07, nik600 [EMAIL PROTECTED] wrote: Hi i am experiencing this problem: MEETME_RECORDINGFILE=/data/asterisk_data/_${DATETIME}_CONFERENCE exten = ,1,MeetMe(666|1Arxq) exten = 9998,1,MeetMe(666|1Axq) exten = 9997,1,MeetMe(666|1xq) I make a conference between 3 person dialing A

Re: [asterisk-users] function call out of AGI script

The documentation says that the exec() command exists to execute applications, not functions. How would I convert an dialplan extension like exten = 11,1,Set(count_naptr=${ENUMLOOKUP(4961369993473,ALL,c)}) into an exec call like $AGI-exec($app, $options) ? On 18/01/07, William Piper [EMAIL

[asterisk-users] CPU Bandwidth Consumption

Is the Asterisk processing and mixing of SIP channels into a single call (simple/minimum, no transcoding etc) calculated in integer or floating point instructions? How much CPU bandwidth is used per call leg, in either MIPS or MFLOPS? How about the G.729 codec, or other codecs:

Re: [asterisk-users] Delay in Call Distribution using the Queue Application

On Jan 15, 2007, at 2:22 PM, [EMAIL PROTECTED] wrote: Hello all, we're using asterisk 1.2.12.1 in an Inbound callcenter using the queue application. If there are many calls in the queue, it sometimes takes up to 30 Seconds before a call is distributed to an agent. For example there are

[asterisk-users] Announce option for meetme - is it used?

Announce option for meetme - is it used? It makes a caller record their name, but I do not see where this name recording is ever used.   Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Fax. 248-836-5101 www.hirotecamerica.com Board member of

Re: [asterisk-users] Sip Phone CID

check out rpid Mark Johnson wrote: Rob Schall wrote: This might sound like an odd question but here it is anyways... We currently have Polycom 501 phones. We have Asterisk with Realtime/MySQL running for our SIP/Voicemail/Extensions. When one phone dials another, the receiving end does

Re: [asterisk-users] Announce option for meetme - is it used?

It is played to the conference/meetme room prior to the user entering, at least in our setup it works. The caller does not here their own announcement. On 1/19/07, BerkHolz, Steven [EMAIL PROTECTED] wrote: Announce option for meetme - is it used? It makes a caller record their name, but I do

Re: [asterisk-users] Sip Phone CID

I set both the trustrpid and sendrpid to yes, but the calling phone still doesn't show the caller id of the person they are calling. Jason Fuermann wrote: check out rpid Mark Johnson wrote: Rob Schall wrote: This might sound like an odd question but here it is anyways... We

Re: [asterisk-users] connecting a FXS-to-sip 4 port device to an avaya system

analog station ports = fxs analog line ports = fxo, assuming 2 wire loop start On Jan 18, 2007, at 8:26 PM, Erick Perez wrote: Thanks Jerry. Are the avaya station ports a special type ? On 1/18/07, Jerry Jones [EMAIL PROTECTED] wrote: Connect to the avaya line ports, not station ports.

Re: [asterisk-users] Asterisk + Unicall + Telmex E1 MFC/R2 Argentina + Meridian

Similar probles I had were fixed incrementing one of the timers, but if you have already done that, I have no idea of your problem, you require to debug the problem and try to find some consistence in the failures, find if the failure is on the Asterisk - telco Link, or in the Asterisk - meridian

[asterisk-users] Integrating asterisk with Toshiba Astrata DK380

Deat all, I am in middle of integrate Asterisk with Toshiba astrata legacy pbx. Following is my setup *Asterisk - Digium TE110P - E1 card in toshiba pbx - Toshiba PBX* A = B C D Asterisk PBX and strata

Re: [asterisk-users] Sip Phone CID

try actually setting the rpid in the dialplan using sipcalledrpid(name,number) Rob Schall wrote: I set both the trustrpid and sendrpid to yes, but the calling phone still doesn't show the caller id of the person they are calling. Jason Fuermann wrote: check out rpid Mark Johnson wrote:

Re: [asterisk-users] How to limit IAX calls

Aaah... I'll try that... A simple variable would be so much easier... :-) BarZ Jonathan k. Creasy wrote: A demonstration: exten = _X.,1,Set(GROUP()=${CALLERID(num)) exten = _X.,n,Set(CDR(AccountCode)=${CALLERID(num)) exten = _X.,n,GotoIf($[${GROUP_COUNT(${CALLERID(num))}

[asterisk-users] Set Parameter of Call Files

Hi all, I'm implementing call files and everything works nicely except that the variable that I set in the call file does not seem to get populated. Channel:SIP/MyProvider/910555 MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context:myCallFileContext Extension:

Re: [asterisk-users] 1 phone 2 voicemail accounts

Hi all, I have a similar issue. I am looking for a way to make 1 phone to subscribe to 2 voicemail accounts on 2 different Asterisk machines on the same LAN linked over IAX2. Management requested that User1 on Asterisk1 should be able to forward a voicemail message to User2 on Asterisk2. All

Re: [asterisk-users] Delay in Call Distribution using the Queue Application

Yes, I confirm the autofill option is present in 1.4, but must be enabled manually not to break compatibility with 1.2. l. On Fri, 19 Jan 2007 15:32:32 +0100, Tom Rymes [EMAIL PROTECTED] wrote: You may be running into the limitation in Asterisk 1.2 (It's fixed in 1.4, I think

Re: [asterisk-users] NAT solutions

Bernardo Vieira wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Gordon Henderson wrote: If you only have one * box behind the NAT gateway then I don't really see a big issue with it to be honest. Port-forward on the firewall/router device (5060 and 1 through 2) to the *

[asterisk-users] Red: Sip Phone CID

Here is what I have in my extensions.conf file now. Trustrcid and sendrcid are turned to yes in the conf file. I'm not fully sure how the SIPCalledRPID works though. The example I found seems to try and provide the stuff automatically (id and name), but does the SIPPEER stuff even exist? I think

RE: [asterisk-users] Parked calls and the # key

Has there been a fix released for this? We have just upgraded to 1.4, and this issue still exists, and is a real pain. Thanks in advance. Gregory P. Scasny Golden Technologies, Inc. http://www.golden-tech.com blocked::http://www.golden-tech.com [EMAIL PROTECTED] blocked::mailto:[EMAIL

Re: [asterisk-users] 1 phone 2 voicemail accounts

You cannot forward voicemails between Asterisk servers. Naija Man wrote: Hi all, I have a similar issue. I am looking for a way to make 1 phone to subscribe to 2 voicemail accounts on 2 different Asterisk machines on the same LAN linked over IAX2. Management requested that User1 on

Re: [asterisk-users] NAT solutions

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Bernardo, Just a thought: Try using a different SIP port for one of the extensions, if possible. Bob... Bob, Tanks for the tip. I had actually done that before, as a matte of fact that's the solution I have in place now. The thing is, even

RE: [asterisk-users] 1 phone 2 voicemail accounts

A little php and SCP would make this work. You could do a web interface like vmail.cgi. -Original Message- From: Eric ManxPower Wieling [mailto:[EMAIL PROTECTED] Sent: Friday, January 19, 2007 9:12 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

RE: [asterisk-users] Integrating asterisk with Toshiba Astrata DK380

Hi, your zapata.con has 'context=from-pstn' Try changing this to 'context=from-zaptel' Robert Jenkins. _ From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Vidura Senadeera Sent: 19 January 2007 15:19 To: asterisk-users@lists.digium.com Cc: [EMAIL PROTECTED] Subject:

[asterisk-users] Disconnect Supervision UK / BT solution?

Hi all I'm using sangoma a200 cards in the UK and have the ongoing, often noted problem of disconnect supervision with BT POTS lines. Just noticed this post on http://www.voip-info.org/wiki/view/UK+Asterisk+Details stating that potentially someone's got a solution : TDM400P amp; Not Detecting

[asterisk-users] Open Source Hosted PBX

Does anyone know if there exists an Open Source Hosted PBX platform based on asterisk? Regards, David ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

Re: [asterisk-users] Queue and Interface time out

Does anyone have ringinuse=no and autopause=yes working together in queues.conf? We assign members to our customer service queue from an application based on actions the agents take on their PCs. No static agents are defined in agents.conf and no members are specified in queues.conf. All

Re: [asterisk-users] Red: Sip Phone CID

I'm not sure about the sippeer stuff, or where they get that variable. We lookup our info in a database to set it. Also to use sipcalledrpid you'll probably need the patch at http://bugs2.digium.com/view.php?id=6643 . Rob Schall wrote: Here is what I have in my extensions.conf file now.

Re: [asterisk-users] NAT solutions

Bernardo Vieira wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Bernardo, Just a thought: Try using a different SIP port for one of the extensions, if possible. Bob... Bob, Tanks for the tip. I had actually done that before, as a matte of fact that's the solution I have in

Re: [asterisk-users] Parked calls and the # key

A fix was posted to the svn version of 1.2 and 1.4. The actual revision number I do not recall. On 1/19/07, Greg Scasny [EMAIL PROTECTED] wrote: Has there been a fix released for this? We have just upgraded to 1.4, and this issue still exists, and is a real pain. Thanks in advance.

[asterisk-users] Re: Background + Dial

Hmm...I have just noticed this issue as well I want the background command to play soundfiles while the dialplan moves on and is dialing a number of zap channels etc... It plays, but essentially ends up being no different than Playback() I note now before posting this, that the Background

[asterisk-users] mISDN

Hi all, i downloaded and installed mISDN with 2.6.8 kernel, but when i try mISDN-init scan (or config) i get this error: [!!] FATAL: bc not in path, please install. Anyone can help me. Tnx Giordano -- No virus found in this outgoing message. Checked by AVG Free Edition. Version:

[asterisk-users] Re: One way choppy sound

On 2007-01-17 10:29:43 -0800, Yelson Vivas [EMAIL PROTECTED] said: Hi Guys I'm conecting 2 astersk servers using this arquitecture (Ext softphone)==sip==(asterisk 1)iax2 trunk(asterisk 2) ===alaw==(pstn) If i call from the Ext to the asterisk 2 the sound is perfect, but if i call

Re: [asterisk-users] Parked calls and the # key

Bruce Reeves wrote: A fix was posted to the svn version of 1.2 and 1.4. The actual revision number I do not recall. Bruce, Have you had a chance to test? I've been looking for the last hour for the bug report I thought you posted earlier this week, but haven't found anything. Doug --

Re: [asterisk-users] mISDN

On Fri, 2007-01-19 at 18:51 +0100, Giordano Grandis wrote: Hi all, i downloaded and installed mISDN with 2.6.8 kernel, but when i try mISDN-init scan (or config) i get this error: [!!] FATAL: bc not in path, please install. Anyone can help me. # yum install bc Regards, Patrick

Re: [asterisk-users] Disconnect Supervision UK / BT solution?

Does anyone have any thoughts/confirmation about this finally being a viable solution? This disconnect supervision problem has plagued TDM and Sangoma cards for a long time! Just to be clear, what is the exact disconnect problem that you see? I have three TDM cards in two different

Re: [asterisk-users] Parked calls and the # key

Here is the bug: http://bugs.digium.com/view.php?id=8804 I have not tried the svn version yet, I modified my features.c file as noted in the bug and have been running it successfully. On 1/19/07, Doug Lytle [EMAIL PROTECTED] wrote: Bruce Reeves wrote: A fix was posted to the svn version of

Re: [asterisk-users] mISDN

Friday, January 19, 2007, 7:06:40 PM, Patrick wrote: On Fri, 2007-01-19 at 18:51 +0100, Giordano Grandis wrote: Hi all, i downloaded and installed mISDN with 2.6.8 kernel, but when i try mISDN-init scan (or config) i get this error: [!!] FATAL: bc not in path, please install.

Re: [asterisk-users] Parked calls and the # key

Bruce Reeves wrote: Here is the bug: http://bugs.digium.com/view.php?id=8804 Thank you very much! Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety.

RE: [asterisk-users] Parked calls and the # key

Yesthanks a billion. I am trying your patch on our 1.4 install right now Thanks again Greg Gregory P. Scasny Golden Technologies, Inc. http://www.golden-tech.com [EMAIL PROTECTED] 219-462-7200 - Ph. 574-233-1300 - Ph. (866) 806-7127 - Toll Free 219-462-7257 - Fax.

[asterisk-users] MIT Using Asterisk - VM Server

I saw this and laughed, the toilet server, AKA the voice mail server is running Asterisk. http://www.networkworld.com/news/2007/011907-mit-your-take.html?page=1 -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have

RE: [asterisk-users] Parked calls and the # key

Worked like a champ Thanks again... Gregory P. Scasny Golden Technologies, Inc. http://www.golden-tech.com blocked::http://www.golden-tech.com [EMAIL PROTECTED] blocked::mailto:[EMAIL PROTECTED] 219-462-7200 - Ph. 574-233-1300 - Ph. (866) 806-7127 - Toll Free 219-462-7257 -

Re: [asterisk-users] Parked calls and the # key

Bruce Reeves wrote: Here is the bug: http://bugs.digium.com/view.php?id=8804 I have not tried the svn version yet, I modified my features.c file as noted in the bug and have been running it successfully. Looks like I'll have to manually make the changes as well. The current 1.2 Branch

RE: [asterisk-users] Queue and Interface time out

-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James Fromm Sent: Friday, January 19, 2007 12:22 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Queue and Interface time out Does anyone have

[asterisk-users] how can PRI, BRI and analog cards achieve a synchronous clock / timing

hello list, i have a problem regarding the synchronisity (clock source) when using multiple cards. e.g. when having connected one PRI port of our TE410P to the telco, i need to have the analog card like the TDM400P or a B410P synchronous to the clock of our telco provider. otherwise faxing on

RE: [asterisk-users] how can PRI, BRI and analog cards achieve a synchronous clock / timing

If you use a channel bank like the Adtran Atlas 550, you can specify a primary sync to the telco, and every subsequent connection to the Atlas uses that sync as a timing source. Expensive, but I expect you can pick one up or something like it on Ebay. Nothin beats an Atlas, though. -Original

[asterisk-users] Ebay Unwired Buyer, Using Asterisk?

Just got a call from Ebay's unwired buyer and The Voice is Allison Smith. Adoption is wide but who is willing to give away their competitive edge (although ebay doesn't really have any real competition). Thanks, Steve ___ --Bandwidth and Colocation

Re: [asterisk-users] Ebay Unwired Buyer, Using Asterisk?

Allison is not exclusively working for asterisk, she also does other recordings. Zao Steve Totaro wrote: Just got a call from Ebay's unwired buyer and The Voice is Allison Smith. Adoption is wide but who is willing to give away their competitive edge (although ebay doesn't really have any

RE: [asterisk-users] Ebay Unwired Buyer, Using Asterisk?

If you think that constitutes a competitive edge then there isn't much help. Personally I think it's great that everyone can leverage off a 'shared' resource like Allison. At least you know it will sound professional and be recorded at a high standard. Regards, Dean Collins Cognation Pty

[asterisk-users] Set(X=10|g) vs Set(GLOBAL(X)=10)

Hi, show application Set says: ---cut--- Set(name1=value1|name2=value2|..[|options]) [...] g - Set variable globally instead of on the channel ---cut--- But someone told me that the proper way is to use Set(GLOBAL(X)=10). Is Set(X=10|g) somewhat deprecated or not? I know that both run

Re: [asterisk-users] Ebay Unwired Buyer, Using Asterisk?

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On 19 Jan 2007, at 21:07, Steve Totaro wrote: Just got a call from Ebay's unwired buyer and The Voice is Allison Smith. Adoption is wide but who is willing to give away their competitive edge (although ebay doesn't really have any real

Re: [asterisk-users] Ebay Unwired Buyer, Using Asterisk?

Dean Collins wrote: help. Personally I think it's great that everyone can leverage off a 'shared' resource like Allison. Now that's a nice wording. Regards, Philipp -- amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de Let's use IT to solve problems and not to create

Re: [asterisk-users] J1/INS1500 and the Redirect Number

and what if you do: zap debug span 2 OR: zap intense debug span 2 Is the value anywhere there? On 1/19/07, Gary Mensenares [EMAIL PROTECTED] wrote: Hello Andrew! Thanks for taking the time to reply. Sorry but no, it doesnt seem to show up. Here's my dial plan: exten = _X.,s,Answer()

[asterisk-users] Re: Audiocodes GPL

Ok, so why was the GPL license violated? Or did Mr. Torvalds offer you an alternate license arrangement? On 1/17/07, Evan Kirstel [EMAIL PROTECTED] wrote: Sorry for the confusion..the MP202 is running Linux! -Original Message- From: Andrew Joakimsen [mailto:[EMAIL PROTECTED] Sent:

Re: [asterisk-users] mISDN

Ciao Giordano, Hi all, i downloaded and installed mISDN with 2.6.8 kernel, but when i try mISDN-init scan (or config) i get this error: [!!] FATAL: bc not in path, please install. Anyone can help me. You have to install the bc package, according to your distribution's package

[asterisk-users] using the Manager to connect caller to conference

Is there a way, via the Asterisk Manager, to dial an extension and connect it with an existing meetme conference? I'm trying to pull callers into a conference as other conference members leave. Thanks in advance. -George ___ --Bandwidth and

Re: [asterisk-users] Red: Sip Phone CID

Jason Fuermann wrote: I'm not sure about the sippeer stuff, or where they get that variable. We lookup our info in a database to set it. Also to use sipcalledrpid you'll probably need the patch at http://bugs2.digium.com/view.php?id=6643 . I looked at this in the past and never made it work

Re: [asterisk-users] Re: Audiocodes GPL

On 12/4/06, Yaniv Nizan [EMAIL PROTECTED] wrote: 2) What does one go about doing to correct GPL violations? Perhaps someone has a generic legal letter that can be used in these situations? This should help answer that question:

Re: [asterisk-users] Set Parameter of Call Files

Lee Jenkins wrote: Hi all, I'm implementing call files and everything works nicely except that the variable that I set in the call file does not seem to get populated. Channel:SIP/MyProvider/910555 MaxRetries: 1 RetryTime: 60 WaitTime: 30 Context: myCallFileContext

Re: [asterisk-users] Asterisk Manager and Ruby

(Sorry for the way-late response to this short thread...) We use rami in production on an Asterisk 1.2.3 server, and have had basically zero problems at least since 1.2.3 was released. rami and ruby's built in RPC provide a very easy to use proxy, if you have multiple clients which all need

RE: [asterisk-users] RE: Polycom buddies question

Hints in extensions.conf in conjuction with mac-directory.xml with bw set to 1. Bill -Original Message- From: [EMAIL PROTECTED] on behalf of Doug Sent: Thu 1/18/2007 1:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion; Asterisk Users Mailing List - Non-Commercial

RE: [asterisk-users] RE: Polycom buddies question

yes it shows the normal Buddies screen that is available from the LCD if that feature is enabled in the Polycom sip config file (presence) Bill -Original Message- From: [EMAIL PROTECTED] on behalf of Doug Sent: Thu 1/18/2007 1:00 PM To: Asterisk Users Mailing List - Non-Commercial

Re: [asterisk-users] Red: Sip Phone CID

Yes it should, I'm not running bleeding edge 1.2 but it isn't an older branch either. Mark Johnson wrote: Jason Fuermann wrote: I'm not sure about the sippeer stuff, or where they get that variable. We lookup our info in a database to set it. Also to use sipcalledrpid you'll probably need

[asterisk-users] Incoming SIP line does not display CallerID correctly

Hi all, I've just setup a sip line with Telasip and when they route the calls to my asterisk box, they include an extension along with the context that is defined in sip.conf for that DID. At first, I couldn't figure why they were getting 404 error from my asterisk box, but then figured

[asterisk-users] Re: asterisk-users Digest, Vol 30, Issue 79

Regards, Vidura B. Senadeera. -- next part -- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070119/0dd5e0be/attachment-0001.htm -- Message: 10 Date: Fri, 19 Jan 2007 11:46:57 -0500

Re: [asterisk-users] Queue and Interface time out

That worked. I don't understand what call-limit has to do with this. I set it to 5. Why does that keep the member interface from getting a second call from the Queue application? I would think it would allow the member interface to get up to 5 calls. Watkins, Bradley wrote:

Re: [asterisk-users] using the Manager to connect caller to conference

see Originate manager Action in voip-info.org On 1/19/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Is there a way, via the Asterisk Manager, to dial an extension and connect it with an existing meetme conference? I'm trying to pull callers into a conference as other conference members

[asterisk-users] Anyone know what this warning is about? Nothing in list history about it either..

On inbound calls from my SIP provider I get multiple warnings as follows: WARNING[5351]: chan_sip.c:7086 check_via: 'MODH6QD' is not a valid host Everything else works but these warnings are a pain and I don't know what they are about Nothing on previos lists or Google explains...

[asterisk-users] Asterisk 1.4 and g723

I am setting up Asterisk for use in a low bandwidth environment. As bandwidth is precious and our ATA's support it, the decision was made to use the g723 codec. I have been working on this for a few days and have not been successful. The issue that I am having is garbled noise at the client on

[asterisk-users] chanskype

Hi, I tried the try version of chanskype, however, everytime that I make a call asterisk generate an error Anyone has experience with this? Since I tried to contact the support but they never replied. Thank you ___ --Bandwidth and Colocation provided

[asterisk-users] CAS on Sangoma boards

Dear folks, I would be very thankful if an experienced user can help me out here. I wanna use mfcr2 and unicall library on sangoma boards but so far impossible for me. As Im setting the framing type to CAS (TE_SIGMODE = CAS) on wanpipe I couldnt get the link alarm out (i looped a A102d links)