[Asterisk-Users] Asterisk on amd SERVER

Hi I am going to install asterisk on an AMD server, did any one had problems installing it on an AMD server ? Regards Kani ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options

RE: [Asterisk-Users] Asterisk on amd SERVER

You shouldn't really have any problems with i386 version on the AMD. When centos moved to 4.3 the x86_64 bit version was a mess with trying to install packages. Regards Mark Brooker T: 02 4959 8670 M: 0415 846 865 F: 02 9882 0947 E: [EMAIL PROTECTED] W: http://www.mbit.com.au -Original

Re: [Asterisk-Users] Asterisk on amd SERVER

Hello, No problems:) Cheers, Madhawa Kanishka Somaratne wrote: Hi I am going to install asterisk on an AMD server, did any one had problems installing it on an AMD server ? Regards Kani ___ --Bandwidth and Colocation provided by Easynews.com --

SV: [Asterisk-Users] Polycom 501 - Disable DND feature?

Solved! In sip.cfg: keys key.scrolling.timeout=1 key.IP_500.9.function.prim=Null/ Thanks to Derek for this solution! Regards, Jan Från: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] För [EMAIL PROTECTED] Skickat: den 4 maj 2006 15:46 Till:

Re: [Asterisk-Users] PCI voltage

Giordano Grandis wrote: Hi all, I have to bought a PCI with 4 PRI but on digium site I saw that there a re two different kind (3,3V and 5v). What’s the difference? 33MHz 32 bit PCI slots are 5V. PCI-X slots MAY support 5V and 3.3V depending on the age of the board. My understanding is

Re: AW: [Asterisk-Users] DTMF detection when outgoing call to mobilephones

Marc Scheuffler wrote: Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different mobile network providers. Nothing. There was a bug in various versions of Asterisk when outbound calls were placed using spool files and then could not detect DTMF from the called party. Without

Re: [Asterisk-Users] SIP Phones behind dynamic IPs

Chris Bagnall wrote: Okay, so assuming I've got to drop the re-registration to a much shorter time than the default of every hour, what are the implications of doing so (in terms of network traffic, load on the asterisk box, etc.)? What's the lowest one can reasonably take it? 10 minutes? 1

Re: [Asterisk-Users] ISAC support?

Trond G. Andersen wrote: Has there been done any work to support ISAC ? ISAC is a proprietary codec from Global IP Sound. There will not be any support for it in Asterisk unless GIPS wants to either open-source the codec (not likely) or allow Digium to license it in the same method as the G.729

RE: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones

There was a bug in various versions of Asterisk when outbound calls were placed using spool files and then could not detect DTMF from the called party. Without more details, including the version of Asterisk you are running, it will be difficult to suggest anything to you other than

[Asterisk-Users] DTMF Tones within my Asterisk on all type of Channels

Hi, i am using Asterisk 1.2.6 with Debian Kernel 2.6.8-2-386 and up 2 date Zaptel, Libpri and so on. My Hardware : 4 x SNOM 320 One Wildcard TDM40B 3 x Allnet 7950 1 x Fritz!Box 7050 - working as a ATA I use Asterisk in my Business stable since last year, the problem occurs since 2 or 3

Re: [Asterisk-Users] Help with IRQ conflict between wct2xxp and eth0

Hi Phil, may sound a stupid advice buthave you tried to change PCI slots on your server? Giorgio Phil Menico wrote: I have a conflict problem with the eth0 card and wct2xxp digium board. The PRI can receive calls but my network connection is gone. When I cat /proc/interrupts I get

Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

Ok, I have to agree here. IF my simple fax server log/tiff archive is not enough to satisfy a client that the fax is genuine I would not want them as my customer. I don't care how much money they spend. Business is business and what I do is what I do. There has to be at least a little bit of

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

2006/5/5, Eric ManxPower Wieling [EMAIL PROTECTED]: whether that makes a difference. Should I switch from hostname to IP address in the register string too? If Asterisk has a DNS lookup failure it will never retry that lookup. Well... I think it should be very easy to solve for the

Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

2006/5/5, Steve Underwood [EMAIL PROTECTED]: In most cases, forensic analysis of the audio from another machine would easily show it was a fake. It would lack tell-tale fingerprints of the true path, unless it was done with extreme care. Certainly using exactly the same model of FAX machine that

Re: [Asterisk-Users] Can I recreate a Fax from a recorded file?

Alejandro Vargas wrote: 2006/5/5, Steve Underwood [EMAIL PROTECTED]: In most cases, forensic analysis of the audio from another machine would easily show it was a fake. It would lack tell-tale fingerprints of the true path, unless it was done with extreme care. Certainly using exactly the

AW: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones

Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2 I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf. The detection is not working with call file, manager originate and not with the dial command to the mobile. I have no ideas left. I got it sometimes to

[Asterisk-Users] Re: Asterisk-Users Digest, Vol 22, Issue 26

Hi to all. My asterisk pbx has a tdm400p card with 2 FXO cards on it. I configured the extensions.conf to send all the call incoming from that zap channels to an IVR system. I see in the asterisk CLI the call incoming and the playback of the message custom/myfile but no sound is played on the

Re: [Asterisk-Users] Festival , Cannot hear the words after ,

Hi John, are using enclosing in quotation mark your statement? (festival('text to speech') ) Giorgio John Joseph wrote: Hi I am trying to use festivall with asterisk , I am using RHEL4 , asterisk1.2.7.1 and festival-1.95-beta , I am able to hear the voice form the text file , when I

RE: [Asterisk-Users] Help with IRQ conflict between wct2xxp and eth0

Go into the BIOS, disable every possible device such as floppy controller, usb, serial, parallel, etc. If that doesn't work, move card to another slot. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Giorgio Incantalupo Sent: Friday, May 05, 2006

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

Tom Engleward wrote: --- Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: If Asterisk has a DNS lookup failure it will never retry that lookup. Never meaning until the next reload command is issued, or until the next restart command is issued, or until the next time the OS reboots, or

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREA CHABLE?

On Thursday 04 May 2006 18:44, Colin Anderson wrote: Ah, true dat. However, if quality was crappy believe me my users would let me know. They are salespeople and wholly intolerant of anything that keeps them from yipping on the phone. I also run exactly the same rig at my house backhauled to

Re: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones

Mark Ackroyd wrote: There was a bug in various versions of Asterisk when outbound calls were placed using spool files and then could not detect DTMF from the called party. Without more details, including the version of Asterisk you are running, it will be difficult to suggest anything to you

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

On Fri, 2006-05-05 at 07:47 -0500, Eric ManxPower Wieling wrote: Only if BOTH hosts are on dynamic IPs. If only one of them is on a dynamic IP, then the dynamic one should be registering with the static server. This is why registration was invented. Until I got static IPs for all of them

[Asterisk-Users] Call Hold and Retrieve

Our current PBX allows us to put a call on hold and then anyone in the building can dial #9XXX and pick up that call. I know that I can replicate a similar function by parking. But I would really like to replicate the existing setup. Something about having to train people to hit more than

Re: [Asterisk-Users] number that starts with star on PAP2

I wrote: In the PAP2's setup there are all of these Vertical Service Activation Codes that start with star and Outbound Call Codec Selection Codes, also the setup menu is accessed by pressing star four times, could they be intefering with dialing numbers that start with a star? And is there

[Asterisk-Users] Repost: External voicemail and MWI on internal phone

Hi, I didn't get any response to this posting so thought I'd post again in the hope that anyone who missed the posting the first time may be able to offer some ideas. This problem isn't specific to the particular model of phone - I can see from sip debug that the local extension itself is

[Asterisk-Users] Registering Remote Sipura to Asterisk (both behind firewall)

Can anybody point or provide working configuration how to register Sipura to Asterisk over the Internet. Both Sipura and Asterisk are behind firewalls. I'll be force to use SIP as that is the only protocol that Sipura is using. Do I need to enter any STUN Server: setting in SIP tab. On

Re: [Asterisk-Users] number that starts with star on PAP2

Why I did to mine is modify all the internal Vertical Service Activation Codes to be **x instead of *x. There is probably a better way, but this worked for me. We tried that, but users report they are still having the same problem (the site is located in a different country so I can't

[Asterisk-Users] Dumping queue_log to MySQL

Anyone have a working solution for this? I played with the demo that came with QueueMetrics to see how they were doing it and it was working for a bit but now somehow every night it stopped. Perl and Tail are still running on the server but the information is not dumping to the MySQL

[Asterisk-Users] problem g729

Hello , Im have this problem before copy codec in the /usr/lib/asterisk/modules before registration .. My asterisk is Asterisk SVN-trunk-r20297 built by xxx@ xxx on a i686 running Linux on 2006-04-20 01:02:07 UTC This erro : codec_g729a.so]May 5 21:39:16 WARNING [6950]:

Re: [Asterisk-Users] problem g729

cristian wrote: My asterisk is Asterisk SVN-trunk-r20297 built by xxx@ xxx on a i686 running Linux on 2006-04-20 01:02:07 UTC This has been covered many times on this list already. The G.729 codec binary is not compatible with current SVN trunk. If you are running SVN trunk in production, stop

Re: [Asterisk-Users] Dumping queue_log to MySQL

We use a Perl script called acd_logger.pl. I can't remember when I got it from now but if you want a copy then mail me directly.RegardsJonJon FarmerTelford, Shropshire, UK- Original Message From: Kevin Savoy [EMAIL PROTECTED]To: Asterisk Users Mailing List - Non-Commercial Discussion

Re: [Asterisk-Users] FW: NuFone Update: DIDs

i got the same email. They emailed everyone later that there was an error. --- Steve Prior [EMAIL PROTECTED] wrote: Rich Adamson wrote: At least outbound calls still work, even though they changed IP addresses (and probably colo locations). Maybe not so much now. I just got

[Asterisk-Users] Call Transfer Disconnect (CT-5)

Hi, Anyone has experience in using Call Transfer Disconnect (CT-5) over a PRI with Asterisk ? Call Transfer Disconnect allows you to transfer a call to a third party and disconnect yourself from the communication and also freeing your PRI channels. Here is a document that explains how

Re: [Asterisk-Users] CARD.XML for MGCP cisco phone

Anyone has a working CARD.XML for cisco MGCP phones? The one on the cisco site is old and it's not working with the new firmwares. Thanks Nico___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

On Friday 05 May 2006 08:47, Eric ManxPower Wieling wrote: Any one of those will work. I disagree; this is NOT an Asterisk DNS issue. I specify my hosts by IP and this still happens (in fact, it happened this morning). -A. ___ --Bandwidth and

[Asterisk-Users] Realtime, 2 server setup problem?

All, we're running realtime and ast 1.2.7.1 stable. The problem I'm having is when you register a SIP device on ServerA, works fine, sip show peer works fine. When you dial the SIP device from ServerB however, it tries to dial, even does a MySQL query like it should but comes back saying no route

Re: [Asterisk-Users] Dumping queue_log to MySQL

On 5/5/06, Jon Farmer [EMAIL PROTECTED] wrote: We use a Perl script called acd_logger.pl. I can't remember when I got it from now but if you want a copy then mail me directly.Regards Jon Jon FarmerTelford, Shropshire, UK - Original Message From: Kevin Savoy [EMAIL PROTECTED]To:

[Asterisk-Users] asterisk 1.2 hisax teles 16.3 isa

Hi I was using asterisk 1.0.7 until now. For my connection with the PSTN I used a good old isa hisax teles 16.3 isdn card. Now while compiling 1.2 I nothiced that i4l and chan_modem is no longer supported? Is there any way of getting my good old ISA card to work under 1.2? The docs keep saying

Re: [Asterisk-Users] Realtime, 2 server setup problem?

Matt Schulte wrote: Is there something obvious I am missing? I googled this to death and cannot find anyone with a similar issue. I remember running into this issue in the past and quite frankly am unsure if it has *ever* worked. Seriously? This has been discussed many times on this list and

RE: [Asterisk-Users] Dumping queue_log to MySQL

http://forums.digium.com/viewtopic.php?t=4073 I modified it some, but it integrates with my cms and provides realtime call stats. Contact me off list if youve like to see how it works. Wes Baehr Ability Business Computing, Ltd. Office: 330.882.0455 x25 Cell: 330.882.0455 x35

Re: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

Andrew Kohlsmith wrote: On Friday 05 May 2006 08:47, Eric ManxPower Wieling wrote: Any one of those will work. I disagree; this is NOT an Asterisk DNS issue. I specify my hosts by IP and this still happens (in fact, it happened this morning). Then you are experiencing a different problem.

[Asterisk-Users] Registering Remote Sipura to Asterisk (both behind firewall)

Can anybody point or provide working configuration how to register Sipura to Asterisk over the Internet. Both Sipura and Asterisk are behind firewalls. I'll be force to use SIP as that is the only protocol that Sipura is using. Do I need to enter any STUN Server: setting in SIP tab. On

[Asterisk-Users] Problem on Zap Channel with IVR

Hi to all. My asterisk pbx has a tdm400p card with 2 FXO cards on it. I configured the extensions.conf to send all the call incoming from that zap channels to an IVR system. I see in the asterisk CLI the call incoming and the playback of the message custom/myfile but no sound is played on the

[Asterisk-Users] Cisco 7970 running SIP question

Group I have a Cisco 7970 Running the newest SIP image. I'm running Asterisk SVN-trunk-r7498 on 2006-04-30 15:11:39 UTC When I get a call the callerid number show something like [EMAIL PROTECTED] I thought I seen somewhere what that was but I'm unable to find the correct wording when

Re: [Asterisk-Users] Cisco 7970 running SIP question

I don't remember exactly what the reasoning on Cisco's part is of having the IP address on there, but it happens on ours too. It shouldn't cause any problems with making outgoing calls from the directory, it's just annoying to see it pop up. As for the date time settings... this is what we

Re: [Asterisk-Users] Unwanted conference with snom320 and asterisk1.07bristuffed

Title: Messaggio We use a large number of Snom 320's and we have this same problem even with Call join on Xfer set to off. I had not previously linked it decisively to the Snoms, but it sounds like that is likely our issue. We've had to stay at the 4.5 firmware because otherwise we get

Re: [Asterisk-Users] Dumping queue_log to MySQL

Kevin Savoy wrote: Anyone have a working solution for this? I played with the demo that came with QueueMetrics to see how they were doing it and it was working for a bit but now somehow every night it stopped. Perl and Tail are still running on the server but the information is not dumping

RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question

Aaron Yes it is very annoying! Thanks for the date time settings. That worked GREAT!!! Thanks - Eric -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Aaron Daniel Sent: Friday, May 05, 2006 11:33 AM To: Asterisk Users Mailing List - Non-Commercial

[Asterisk-Users] Music on Hold from Soundcard

Hi everyone, Sent this out previously, but it didn't seem to show up. My apologies if this is a duplicate! I've been trying to get MoH to work from the line-in on my soundcard, but as of yet have had no success. I found this script that should allow for it to happen:

Re: [Asterisk-Users] number that starts with star on PAP2

You need to modify the dialplan within the PAP2 unit to allow that as a valid number or it won't pass it on. Take a look at the following, it is not specifically for the PAP2 but all the dialplan information should apply:http://www.sipura.com/Documents/SipuraSPAUserGuidev2.0.9.pdf From: "Time

[Asterisk-Users] Access to sip.conf username field from dialplan

I have two SIP clients defined in sip.conf, as follows: [dave] username=dave type=friend secret=dave host=dynamic context=local accountcode=deskoptional [dave-laptop] username=dave type=friend secret=dave host=dynamic context=local accountcode=deskoptional I want to refer to the

RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question

Aaron Any idea how to change it from 24hr to 12hr ? Thanks again! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Friday, May 05, 2006 11:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: Spam? Re:

RE: Spam? Re: [Asterisk-Users] Cisco 7970 running SIP question

The A at the end of the dateTemplate sets that. Should read M/D/YA instead of M/D/Y. Aaron On Fri, 5 May 2006, Hall, Eric M. wrote: Aaron Any idea how to change it from 24hr to 12hr ? Thanks again! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of

RE: [Asterisk-Users] FW: NuFone Update: DIDs

I didn't get that message, but I got a we're sorry for sending out the bogus messages message, saying that it was an error.. -Steve -Original Message- From: Steve Prior [mailto:[EMAIL PROTECTED] Sent: Thursday, May 04, 2006 6:40 PM To: [EMAIL PROTECTED]; Asterisk Users Mailing List -

RE: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREACHABLE?

Just as another datapoint, I have a cheap (~$17) walmart cordless phone at home hooked to my digium dual FXS card, and it works great, with the possible exception that there is a buzz (I perceive it as ground hum) for about the first 4 seconds of EVERY call, and slowly it diminishes. I don't know

[Asterisk-Users] AASTRA 9133i and PIX Firewall

Hi, Does anyone have experience with the Aastra 9133i and a PIX firewall? My problem is that it passes data with no problem at all (that being audio goes fine)... however my voicemail notifications do not work. If I take the phone OUT from behind the PIX firewall, voicemail notifications work.

Re: [Asterisk-Users] Registering Remote Sipura to Asterisk (both behind firewall)

Hi, you have to forward port 5060 udp and all the others udp ports that asterisk uses for RTP, 1 to 2 by default, see rtp.conf. At the sipura device in the server configuration you have to put the LAN ip of your Asterisk box as sip proxy server and firewall's ip as out-bound sip proxy.

Re: [Asterisk-Users] Registering Remote Sipura to Asterisk (both behind firewall)

On Fri, 2006-05-05 at 15:37 -0300, Lucas Alvarez wrote: Hi, you have to forward port 5060 udp and all the others udp ports that asterisk uses for RTP, 1 to 2 by default, see rtp.conf. Thank you for the hint. The above part is clear. At the sipura device in the server configuration

[Asterisk-Users] Call Center Phone with Auto Answer

Can anyone recommend a phone to use in an inbound call center environment that has an auto answer feature? We dont want the agents having to acknowledge the call. The call should just activate on the headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601. None of these

[Asterisk-Users] Passing SIP Subscriptions???

Can anyone tell me if they know if it's possible to pass/copy sip subscriptions from one Asterisk system to another? Can IAX do this? What about regexten? Doug. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing

RE: [Asterisk-Users] Call Center Phone with Auto Answer

The Snom 360 spec says it has an auto-answer feature. You can download the data sheet here http://www.snom.com/snom360_voip_phone.html Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225

Re: [Asterisk-Users] Call Center Phone with Auto Answer

Kevin Savoy wrote: Can anyone recommend a phone to use in an inbound call center environment that has an auto answer feature? We don’t want the agents having to acknowledge the call. The call should just activate on the headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601.

Re: [Asterisk-Users] Call Center Phone with Auto Answer

Kevin Savoy wrote: Can anyone recommend a phone to use in an inbound call center environment that has an auto answer feature? We don’t want the agents having to acknowledge the call. The call should just activate on the headphones. We have tried Grandstream 2000, Polycom 301, 501 and 601.

RE: [Asterisk-Users] Call Center Phone with Auto Answer

The problem with what is in wiki is that these calls are being sent to a queue. There is no way to have the queue dial the preceding digit that I can think of that would trigger this. In the example shown he has an 8 dialed before the extension. How would I get Asterisk to dial an 8 before sending

RE: [Asterisk-Users] Call Center Phone with Auto Answer

We are using agent login but we don't want MOH on the line at all times as some of these phones could and probably will be connected in remote locations. We don't want to stream MOH across frames chewing up bandwidth when there are no calls to present to that phone. We do have the ackcall=no in

Re: [Asterisk-Users] Cisco 7970 running SIP question

I don't remember exactly what the reasoning on Cisco's part is of having the IP address on there, but it happens on ours too. It shouldn't cause any problems with making outgoing calls from the directory, it's just annoying to see it pop up. It's so the phone routes the call to the correct

[Asterisk-Users] Code parsing error?

This code executes just fine, and leaves the SIP peer's mailbox setting from sip.conf in variable target. exten = 1,1,Set(target=${CHANNEL:4}-) exten = 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox}) exten = 1,n,VoiceMailMain(${target}) However, every time it runs I get an error

[Asterisk-Users] 300 DID's required in Alpine Texas Area code 432

Have customer who requires 300 DID'd in Alpine Tx 432 if the price is right. Email with details. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit:

[Asterisk-Users] Asterisk -- NAT -- Internet -- NAT -- Sipura-3K (No Asterisk)

So far I have gathered that on my NAT (where asterisk server is) I have to port forward UDP ports: 5060 and range 1 - 2 to my asterisk server But I'm still stuck how to configure Sipura (behind NAT) what sip proxy and out-bound sip proxy, under which tab to change. -- #Joseph

[Asterisk-Users] Code parsing error?

Did you mean to have the dash inside the braces -- this may be your problem. on Friday 05/05/2006 David L. West([EMAIL PROTECTED]) wrote This code executes just fine, and leaves the SIP peer's mailbox setting from sip.conf in variable target. exten = 1,1,Set(target=${CHANNEL:4}-)

[Asterisk-Users] Bandwidth via my Asterisk PBX

Am new to Asterisk - have it up and running connected to a couple service providers (telasip teliax). Nice! Am running our Asterisk PBX on a static DSL connection (~600 kbps up/~4mbps down), and would like to extend VoIP service to 10 non-profits we're working with. Am I correct in assuming

[Asterisk-Users] asterisk behind load-balancing switch

Hi, I have 2 SER and 2 Asterisk boxes behind a load-balancing switch. I need Asterisk to initiate the RTP streams to both endpoints. Can that be done? Right now, Asterisk does it part of the time, so I have to create NATs for the Asterisk boxes for the RTP streams to get in. Thanks.

[Asterisk-Users] ODBC Voicemail storage and app_directory

I've checked the wiki, searched the mailing list, Mantis (bug tracker), and looked through the source code, but found not mention of this issue: If using ODBC Storage for voicemail messages, name-playback by app_directory breaks. app_directory has no ODBC Storage code and therefore only looks

RE: [Asterisk-Users] why a perfectly fine iax2 host becomes UNREA CHABLE?

toughest asterisk critic (my wife) yes, my wife is worth more in real-world this sucks feedback than an entire office. keeps me on my toes. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or

[Asterisk-Users] Passing Callerid

How do I setup a dial plan that will allow a customer to set their own callerid? No matter what I try, the username that I have for them in sip.conf is passed because I have no callerid setup for them. If I setup a callerid in sip.conf that is passed I can change it and it works. That is well

[Asterisk-Users] Running Asterisk as non-root

I saw where one should not run Asterisk as root: http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root How important is this? Thanks, just curious whether it's worth the trouble. H ___ --Bandwidth and Colocation provided by Easynews.com --

Re: [Asterisk-Users] Running Asterisk as non-root

I saw where one should not run Asterisk as root: How important is this? It's probably not very important at the moment, however, it's not that hard to do either. I run Asterisk non-root and in a chrooted environment -- it keeps all necessary files nicely separated (easily portable, easy to

RE: [Asterisk-Users] Running Asterisk as non-root

If you run Asterisk as non root, you may have problems installing G729 licenses. The digium registration utility has certain hard coded stuff, and doesn't behave well when things aren't installed in the standard location. Doug. -Original Message- From: Luki [mailto:[EMAIL PROTECTED]

[Asterisk-Users] Re: Running Asterisk as non-root

Thanks for the swift response Luki. Have you checked out: http://www.voip-info.org/wiki/index.php?page=Asterisk+non-root recently? Does it seem up to date to you? It indicates it was updated in March of this tear, but I did this once and don't want to go thru all the hassle if the directions

[Asterisk-Users] Re: Running Asterisk as non-root

Oh boy - thanks. I use g729... H On 5/5/06, Douglas Garstang [EMAIL PROTECTED] wrote: If you run Asterisk as non root, you may have problems installing G729 licenses. The digium registration utility has certain hard coded stuff, and doesn't behave well when things aren't installed in the

[Asterisk-Users] REGISTER that isn't a register

Anyone come across this message: WARNING[2203]: chan_sip.c:9633 handle_response_register: Got 200 OK on REGISTER that isn't a register I get it from time to time but don't know what to do about it! Thanks, h ___ --Bandwidth and Colocation provided by

[Asterisk-Users] ASTERISK DISA FOR INCOMING DID CALL

Hi, I am trying to create a situation where I call the DID number which is 1140636249 and I receive a dial tone to dial. I d like also to autenticate the number 1130851536. I can see that asterisk decode this number but I dont know how to authenticate this number only. This is what I

Re: [Asterisk-Users] Code parsing error?

At 02:06 PM 5/5/2006, you wrote: exten = 1,1,Set(target=${CHANNEL:4}-) However, every time it runs I get an error in the CLI as follows WARNING[5629]: pbx.c:1366 ast_func_read: Can't find trailing parenthesis? This happens right after it executes the first line of code, then

Re: [Asterisk-Users] Re: Running Asterisk as non-root

If you run Asterisk as non root, you may have problems installing G729 licenses. The digium registration utility has certain hard coded stuff, and doesn't behave well when things aren't installed in the standard location. Good point. However, in the chrooted environment there is no need to

Re: [Asterisk-Users] Code parsing error?

On Fri, May 05, 2006 at 03:06:43PM -0600, David L. West spake thusly: This code executes just fine, and leaves the SIP peer's mailbox setting from sip.conf in variable target. exten = 1,1,Set(target=${CHANNEL:4}-) exten = 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox})

RE: [Asterisk-Users] Call Center Phone with Auto Answer

I don't see any reason you can't use a polycom. You should be able to solve your problem multiple ways. You can simply put the default ring on the Polycom to autoanswer if that is the sole purpose, give it a second extension to be used in the queue that is programmed to autoanswer, as a couple of

[Asterisk-Users] Silent Attendant

I'd like to set up a silent attendant. By this I mean when someone calls me I'd like for them to hear the comfort ringing tones, but for the first 5 seconds I'd like to give them the option of pressing 9 to send the call to an alternate extension; if they don't press 9, the call goes to a

[Asterisk-Users] Info

Hi all, anyone could pls explain me what does it means ? It a part of zaptel.conf file. LBO= Line Build Out 0:0dB(CSU)/0-133feet(DSX-1) 1:133-266feet(DSX-1) 2:266-299feet(DSX-1) 3:399-533feet(DSX-1) 4:533-655feet(DSX-1) 5:-7.5dB(CSU) 6:-15dB(CSU) 7:-22.5dB(CSU) Thanks

[Asterisk-Users] How to determine if a device is in a call

I have gotten intercom working on my office phones (Linksys SPA-942s), but I have noticed that if someone is in a call, it places the call on hold and sends the intercom audio to the person holding the phone that is being paged. I'd like to add logic to my dialplan that doesn't send a page to a

[Asterisk-Users] CW options not changing

I have a weird issue with a system, running freepbx in devices and users mode (same as dozens of systems) but this one when you hit *70 and get "call waiting activated" it is not storing the setting in the database. I can manually set CW 900 ENABLED but then *71 does not disable it. Any

Re: [Asterisk-Users] Re: Running Asterisk as non-root

At 04:22 PM 5/5/2006, you wrote: I don't have time to write up the steps to chroot asterisk, but if anyone is interested then I will tonight. Personally, I'd be very interested. Thanks so much, Ira ___ --Bandwidth and Colocation provided by

[Asterisk-Users] Re: Code parsing error?

There's definitely a ) missing in this line! A good text editor with Thanks( I finally spotted it after fortifying myself with a good Thai dinner). I've been using gEdit, so should probably start shopping for something vim-ish. ___ --Bandwidth

Re: [Asterisk-Users] How to determine if a device is in a call

Check in voip-info.org about sip.conf parameter call-limit. It may help you. Regards On 5/5/06, Carl Youngblood [EMAIL PROTECTED] wrote: I have gotten intercom working on my office phones (Linksys SPA-942s), but I have noticed that if someone is in a call, it places the call on hold and sends

RE: [Asterisk-Users] How to determine if a device is in a call

Show application chanisavail Should be what you need Wes Baehr Ability Business Computing, Ltd. Office: 330.882.0455 x25 Cell: 330.882.0455 x35 [EMAIL PROTECTED] -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Carl Youngblood Sent: Friday, May 05,

[Asterisk-Users] Multiple periodic announcements in queues? Possible?

Hi all! Curious if there was a way to introduce multiple announcments in the queues. Rather than just: Thank you for holding. Your call is important to us ...wait 60 sec... [repeat] Is it possible to: Thank you for holding. Your call is important. ...wait 60 sec... Did you know

RE: [Asterisk-Users] Silent Attendant

Simply generate or record a 5-second sample of ringing. Then use Background() to play that ringing file - if someone presses 9, they will be routed accordingly, or otherwise sent to your default extension. Wes Baehr Ability Business Computing, Ltd. Office: 330.882.0455 x25 Cell: 330.882.0455

[Asterisk-Users] Re: Code parsing error?

Oh, on the off chance that it's interesting to anybody what I'm actually doing here. I have two SIP devices, "dave" and "dave-laptop". I want to have one voicemail box but be able to check it from either one. In sip.conf, both devices have "mailbox=dave". It's this setting that I'm

RE: [Asterisk-Users] Silent Attendant

I changed my mind. This worked for me, using AEL: Context test { 825 = { Set(TIMEOUT(response)=5); Answer(); Ringing(); WaitExten(5); Goto(menu-main|s|1); }; }; Or in regular format: exten =

Re: [Asterisk-Users] Re: Code parsing error?

Title: Re: [Asterisk-Users] Re: Code parsing error? On 5/5/06 10:45 PM, David L. West wisely said: exten = 1,n,Set(target=${SIPPEER(${CUT(target,,1)}:mailbox}) Well: Exten = 1,n,Set(target=${SIPPEER(${CUT(target,,1)}):mailbox}) It looks like you didnt end the parenthesis for SIPPEER().

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