Re: [asterisk-users] call transfer detection in dial plan

On 9/13/07, Rilawich Ango <[EMAIL PROTECTED]> wrote: > Hi all, > In default, we can use # to transfer the call. I want to know how I > can know the user presse # to transfer the call in dial plan. > ango Set TRANSFER_CONTEXT or GOTO_ON_BLINDXFER variable (depending on * version) before Dial().

Re: [asterisk-users] Agent Callback Login in 1.4

On Wed, 2007-09-12 at 21:45 -0700, Kevin P. Fleming wrote: > Paul Hales wrote: > > > I have written stuff using the addqueuemember, but you lose agent level > > functionality and reporting. :( > > Can you describe exactly what you lose by using the dynamic queue member > alternative? We tried to

Re: [asterisk-users] asterisk call back dail plan

Hi, I meant - if you have more specific questions - please ask them. And writing back to ML would be desirable, because this info might be useful for other people. I can't give you my dialplan, because it's too large and probably useless without lot of external configs. I can just tell you where t

[asterisk-users] Licensing and provisionning

Hello, As we mostly use Open Source software, we have automated process to build new servers from scratch. Unfortunately, this process is somehow broken by having to type by hand licensing data (e.g. HPEC or G729 licences) and it's not so easy to maintain accurate licensing data so that someone c

Re: [asterisk-users] FW: Problems with two trunks

Does the mytel gateway show up fine in sip show peers? PaulH On Thu, 2007-09-13 at 15:06 +1000, Joshua Small wrote: > You can ignore this. I mistyped the password, and once it was fixed, > and registered correctly, both links failed to work again. > > I have some extended information from sip

[asterisk-users] FW: Problems with two trunks

One more spam from yours truely.. Got it sorted out. My configuration never had a: Fromuser = 111 In it. Worked fine for any individual trunk. But after sticking that command in for each individual connection, I was able to register two of them (and use them both). Thanks for th

[asterisk-users] FW: Problems with two trunks

You can ignore this. I mistyped the password, and once it was fixed, and registered correctly, both links failed to work again. I have some extended information from sip debug. Again, this shows up as soon as I try to register two connections. <--- SIP read from 203.166.103.242:5060 ---> SIP/

Re: [asterisk-users] Agent Callback Login in 1.4

Paul Hales wrote: > I have written stuff using the addqueuemember, but you lose agent level > functionality and reporting. :( Can you describe exactly what you lose by using the dynamic queue member alternative? We tried to ensure that no functionality was lost in this transition, so if there is

[asterisk-users] call transfer detection in dial plan

Hi all, In default, we can use # to transfer the call. I want to know how I can know the user presse # to transfer the call in dial plan. ango ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocati

Re: [asterisk-users] Agent Callback Login in 1.4

On Thu, 2007-09-13 at 15:23 +1200, Matt Riddell wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Paul Hales wrote: > > It's a great feature, and one hopes it will return one day. > > > > PaulH > > > > > > On Wed, 2007-09-12 at 19:01 -0700, Ryan Stark wrote: > >> I tried to use it ba

Re: [asterisk-users] bug in 1.2.24

Thank Isaac, Ill try it this way.. Im currently using this before entering the queue so calls from the queue are recorded: exten => s,n,SetVar(MONITOR_FILENAME=/var/spool/asterisk/${TIMESTAMP}-${UNIQUEID}-${C ALLERIDNUM}-Queue-Ventas) exten => s,n,SetVar(TRANSFER_CONTEXT=internalphones) So I coul

[asterisk-users] FW: Problems with two trunks

Update on this: I found that by changing insecure = very to insecure = invite, adding the second trunk no longer stopped calls working. I've read the documentation on this switch and still don't see how it applies/is meant to get used. Anyway, with this change in place, the following may h

Re: [asterisk-users] Trunk & Outbound Route for a Cisco VOIP router?

Doug, Not sure on the trixbox side but for asterisk: Asterisk server: 10.0.0.1 Cisco Gateway: 10.0.0.2 In sip.conf [cisco] context=cisco type=friend host=10.0.0.2 dtmf=rfc2833 extension.conf exten=>_011.,1,Dial(SIP/[EMAIL PROTECTED]) In the Cisco: dial-peer voice 100 voip application session

Re: [asterisk-users] Agent Callback Login in 1.4

Matt Riddell wrote: > -BEGIN PGP SIGNED MESSAGE- > Hash: SHA1 > > Paul Hales wrote: > >> It's a great feature, and one hopes it will return one day. >> >> PaulH >> >> >> On Wed, 2007-09-12 at 19:01 -0700, Ryan Stark wrote: >> >>> I tried to use it back in 1.4.6 or so and it is hor

[asterisk-users] No Sound on Zap Channels

All, I've got a strange issue here. When I make a SIP call to say my voicemail app, I hear audio just fine. However when I dial from PSTN into my Asterisk box, I see that its playing the voice files, but I hear nothing, then the call drops. I'm running Fedora Core 6, and Asterisk 1.2.24. CLI ou

Re: [asterisk-users] Agent Callback Login in 1.4

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Paul Hales wrote: > It's a great feature, and one hopes it will return one day. > > PaulH > > > On Wed, 2007-09-12 at 19:01 -0700, Ryan Stark wrote: >> I tried to use it back in 1.4.6 or so and it is horribly broken, I >> ended up rewriting the func

[asterisk-users] Trunk & Outbound Route for a Cisco VOIP router?

Hi, I am trying to set up TrixBox/FreePBX with a trunk and outbound route to a Cisco VOIP router. Has anyone on the list done this successfully? Willing to share config file snippets? Mucho thanks if you can help! ___ Sign up now for AstriCon 2007!

Re: [asterisk-users] Callback for unanswered transfers...

Thank you. 2007/9/12, Atis <[EMAIL PROTECTED]>: > > On 9/12/07, Luis Antonio Prata Barbosa <[EMAIL PROTECTED]> wrote: > > Hi, > > > > Does anybody know if there is a way for a call goes back to transferer > if > > unanswered ? > > Yes, before Dial to transferer set some variable that have he's > e

Re: [asterisk-users] Agent Callback Login in 1.4

It's a great feature, and one hopes it will return one day. PaulH On Wed, 2007-09-12 at 19:01 -0700, Ryan Stark wrote: > I tried to use it back in 1.4.6 or so and it is horribly broken, I > ended up rewriting the functionality with dynamic queue members in the > dial plan. I really liked the c

Re: [asterisk-users] Conference bridge.

On Wed, 2007-09-12 at 16:44 -0400, Alex Balashov wrote: > Any recommendations for an affordable SIP conference bridge unit? I mean > one that isn't crappy; something where the duplex and cancellation > functions that are traditionally built into such devices actually work. Do you want somethin

Re: [asterisk-users] Agent Callback Login in 1.4

I tried to use it back in 1.4.6 or so and it is horribly broken, I ended up rewriting the functionality with dynamic queue members in the dial plan. I really liked the call back agent feature set. I found it to be far superior to dynamic queue member alternative. -Ryan On 9/12/07, Anthony Franci

Re: [asterisk-users] Agent Callback Login in 1.4

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Anthony Francis wrote: > Awhile back I had heard some talk, in this list I believe that Agent > callback login was going to be deprecated in 1.4, I see it is still > there. Does anyone know what is happening with this? It has been deprecated and use

Re: [asterisk-users] Conference bridge.

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Alex Balashov wrote: > Any recommendations for an affordable SIP conference bridge unit? I mean > one that isn't crappy; something where the duplex and cancellation > functions that are traditionally built into such devices actually work. Most peo

Re: [asterisk-users] Problems with two trunks

I would have usually used sip.conf or iax.conf - users.conf is not something I know well PaulH On Thu, 2007-09-13 at 10:44 +1000, Joshua Small wrote: > Hi, > > > > I am attempting to setup an asterisk server, current specs: > > CentOS release 5 (Final) > > Asterisk 1.4.11 > > Asteris

[asterisk-users] Zap channels: no sound with certain call paths

Hi, A most peculiar and vexing problem for you all. I hope I have been verbose enough without being a firehose ;) The set up: I have a channel bank, using the r1t1 rhino driver with a rhino T1 card (the channel bank itself is a very legacy piece of equipment)- this supplies FXS for all the house p

[asterisk-users] Problems with two trunks

Hi, I am attempting to setup an asterisk server, current specs: CentOS release 5 (Final) Asterisk 1.4.11 Asterisk-gui checked out from SVN last week I started with a fairly basic setup involving one VOIP provider who provided one dial in number, and a couple of handsets. Config files are

Re: [asterisk-users] Generating an old-fashioned dialtone

On Wed, Sep 12, 2007 at 10:13:03AM +0100, Phil Reynolds wrote: > I can probably put it on Zap phones easily enough if I wish, but I'd > need to know how to generate it first, and all I am after right now is > the sound. I believe that's roughly 250hz beating with 10hz. Cheers, -- jra -- Jay

[asterisk-users] Assistance needed.

I am looking for help from someone familiar with using asterisk and openser to build a rather large VOIP network. I have 6 servers in place each with their own purpose. I will give a brief summary and hopefully someone out there is able to help be finalize this dialplan. I have six servers in place

[asterisk-users] AsteriskNOW

Hello, is there any User Interface available in Asterisk NOW? in Trixbox, As far as I know there is ARI, but does Asterisk Now has anything for the Extension owners, I mean User portal not Admin portal? Regards, Seysan ___ Sign up now for AstriCon

Re: [asterisk-users] Flash IDE

On Wed, 2007-09-12 at 09:19 -0400, Jon Pounder wrote: > there is tons of information about linux and flash drives on the > nslu2-linux.org and the openwrt sites. > > main points : > > - disable swap > - disable atime > - disable most logging > > once the drive is not being written to then it w

Re: [asterisk-users] Generating an old-fashioned dialtone

On Wed, Sep 12, 2007 at 02:29:35PM -0500, Anthony Messina wrote: > On Wednesday 12 September 2007 02:57:18 am Phil Reynolds wrote: > > Is there a way to generate an old-fashioned dial tone with Asterisk? > > > > I'm thinking of one that sounds like: > > > > http://www.seg.co.uk/telecomm/dialtone.wa

Re: [asterisk-users] Digium Appliance

Sounds robust to me. David Boyd wrote: > Hi Mat, > i have been working with the aa50 for a couple of weeks now. They are > slick looking devices that still have a few bugs. I tried to use the > device like an end user without previous knowledge of Asterisk or the > asteriskGUI, and can say right

Re: [asterisk-users] Generating an old-fashioned dialtone

On Wed, Sep 12, 2007 at 11:23:51AM -0600, Stephen Bosch wrote: > It's been years since I was in the UK. I can't remember what the modern > dial tone sounds like. When did it change? The first version of it appeared in parts of Sutton Coldfield in 1976, but some places still had the old tone into t

Re: [asterisk-users] Digium Appliance

Hi Mat, i have been working with the aa50 for a couple of weeks now. They are slick looking devices that still have a few bugs. I tried to use the device like an end user without previous knowledge of Asterisk or the asteriskGUI, and can say right off that a typical person will not be able to use

Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

I read the article, and it seems he was talking about cleaning the code and making it the best it could be, while the author was talking about the other things. Anthony shadowym wrote: > > Maybe his comments were taken out of context as they don’t have the > whole interview posted. Why is he t

[asterisk-users] Agent Callback Login in 1.4

Awhile back I had heard some talk, in this list I believe that Agent callback login was going to be deprecated in 1.4, I see it is still there. Does anyone know what is happening with this? -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP (303) 444-7052 opt 2 [EMAIL PROTECT

[asterisk-users] Conference bridge.

Any recommendations for an affordable SIP conference bridge unit? I mean one that isn't crappy; something where the duplex and cancellation functions that are traditionally built into such devices actually work. I am referring to something that looks like this . . . http://www.hardware.com/p

[asterisk-users] Looking for Asterisk Consultant in San Franicsco

Hello, I am looking for an Asterisk consultant for occasional support on an asterisk phone system located in San Francisco. It would probably be primary remote support, but we may need some on site support occasionally. Please let me know if you are interested and available. Thanks, Niki

[asterisk-users] (no subject)

Hello, I am looking for an Asterisk consultant for occasional support on an asterisk phone system located in San Francisco. It would probably be primary remote support, but we may need some on site support occasionally. Please let me know if you are interested and available. Thanks, Niki

Re: [asterisk-users] Generating an old-fashioned dialtone

On Wednesday 12 September 2007 02:57:18 am Phil Reynolds wrote: > Is there a way to generate an old-fashioned dial tone with Asterisk? > > I'm thinking of one that sounds like: > > http://www.seg.co.uk/telecomm/dialtone.wav see if you can find it here http://www.3amsystems.com/wireline/tone-search

Re: [asterisk-users] Callback for unanswered transfers...

On 9/12/07, Luis Antonio Prata Barbosa <[EMAIL PROTECTED]> wrote: > Hi, > > Does anybody know if there is a way for a call goes back to transferer if > unanswered ? Yes, before Dial to transferer set some variable that have he's extension, and in your defined TRANSFER_CONTEXT, use Dial with g opti

Re: [asterisk-users] Different Networks

On 9/7/07, Mike Hammett <[EMAIL PROTECTED]> wrote: > If it has nothing to do with Asterisk, then why does every other device work > as its supposed to? You never answered as to whether or not you're able to get out past your gateway with any other network applications on your asterisk server. Fir

[asterisk-users] Callback for unanswered transfers...

Hi, Does anybody know if there is a way for a call goes back to transferer if unanswered ? Thanks Luis A P Barbosa ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] TE405P intermittent yellow alarm

Richard van der Hoff wrote: > Folks, > > I really hope you can help me here - I'm beginning to tear my hair out! > > About 10 days ago my company moved to a new office. As a result of this, > we've plugged our PBX box, which has happily been running for the last > three years, into our new E1 line.

Re: [asterisk-users] Generating an old-fashioned dialtone

Phil Reynolds wrote: > Quoting Clayton Milos <[EMAIL PROTECTED]>: >>> Is there a way to generate an old-fashioned dial tone with Asterisk? >>> >>> I'm thinking of one that sounds like: >>> >>> http://www.seg.co.uk/telecomm/dial tone.wav > >> As far as I know dialtone with SIP can only be generated

Re: [asterisk-users] online active call watching

Dinesh Nair wrote: > On Mon, 10 Sep 2007 13:43:46 -0800, Mojo with Horan & Company, LLC wrote: > > >> Though still in the proof-of-concept stage, my project "AstSee" from >> http://www.astsee.com/ might be fun to play with if you're using >> linux/XWindows. There are screenshots there. >>

Re: [asterisk-users] Installing Asterisk on to CentOS 4

Ove Aursand wrote: > Anthony Francis wrote: >> Ove Aursand wrote: >> >>> Abdul wrote: >>> Hi expets, I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. But when i am trying to start asterisk with following cmd i am getting unknown c

[asterisk-users] Digium Appliance

Hi, Has anyone actually gotten their hands on an appliance yet? If so, how robust and working are they? Any issues? ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.ap

Re: [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?

I prefer centos , debian/ubuntu are also a good option . It just depends on which distribution you are comfortable with . We also have asterisk running very stable on slackware . On 12/09/2007, Gordon Henderson <[EMAIL PROTECTED]> wrote: > > On Wed, 12 Sep 2007, Euler Pereira wrote: > > > Hey all!

Re: [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?

On Wed, Sep 12, 2007 at 05:25:16PM +0100, Gordon Henderson wrote: > On Wed, 12 Sep 2007, Euler Pereira wrote: > > > Hey all! > > > >I'm newbie in the Asterisk World but old in other telephony systems like > > Lucent/Avaya, Sopho, Siemens and Linux/Unix system. > > > >I'm in doubt, as based

Re: [asterisk-users] Flash IDE

Diego Iastrubni wrote: > Disable DMA on that drive. Thee HD/DOM/CF-card does not support DMA and linux tries to "DMA" it. This is with ide=nodma kernel option. It's just loading the kernel and initrd at the beginning, before the kernel's actually booted. > (gmail quoting stinks) huh? what's

Re: [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?

On Wed, 12 Sep 2007, Euler Pereira wrote: > Hey all! > >I'm newbie in the Asterisk World but old in other telephony systems like > Lucent/Avaya, Sopho, Siemens and Linux/Unix system. > >I'm in doubt, as based system, should I install Fedora, Debian, > Slackware, FreeBSD our Sun Solaris? Wh

Re: [asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?

As far as I know, Asterisk was developed on Linux, and although it will work on other systems, my best experiences have been with Linux. This is especially true once you start talking about integration with Zaptel, since it requires kernel module hooks, etc. And I think your situation does

Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

On 9/12/07, Al lists <[EMAIL PROTECTED]> wrote: > I liked the queue game concept! > although it could be cruel! Hmm, can't find anything about queue games in google. Anybody have some more details? I would like to have some functionality of that.. Regards, Atis -- Atis Lezdins, IT Responsible o

[asterisk-users] Linux Fedora, Debian, Slackware, FreeBSD our Sun Solaris?

Hey all! I'm newbie in the Asterisk World but old in other telephony systems like Lucent/Avaya, Sopho, Siemens and Linux/Unix system. I'm in doubt, as based system, should I install Fedora, Debian, Slackware, FreeBSD our Sun Solaris? Which is more robust for a small Asterisk system, about

[asterisk-users] res_snmp

Hi, I have problems compiling asterisk 1.4.11 with res_snmp. I do 'make menuselect', and I see that this resource module depends on netsnmp. I am using centOS 4.5. I do: > yum install net-snmp net-snmp-devel net-snmp-utils net-snmp-libs I don't know if i am missing something. I go to the source d

[asterisk-users] Solution: Sysmaster and Asterisk

Hello Guys, After adding money into my sysmaster phone account I am able to make calls outside.thnx _ From: Mani Nair [mailto:[EMAIL PROTECTED] Sent: Friday, September 07, 2007 9:16 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Sysmaster and Asterisk

Re: [asterisk-users] Problems with Asterisk behind a firewall

On 2007-09-12 at 13:41 lemmel lemmel wrote: >I don't have enough experiment to help you, but I can suggest you this : >http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html Many thanks, will have a look. Christian > >_

Re: [asterisk-users] Direct dialing to correct extension from analog lines

On Wed, Sep 12, 2007 at 04:55:39PM +0200, Lars Bensmann wrote: > Hi, > > I have a problem with people that are calling from analog lines. > > We have a block of numbers 12345 - 0 to -99. 00 to 99, right? > Most calls are transmitting > the whole number including the extension. There's no probl

Re: [asterisk-users] Astribank 32 and Far End Disconnection

On Wed, Sep 12, 2007 at 11:59:41AM -0300, Gleidson Antonio Henriques wrote: > Hi all, > >I'm interested in buy a Astribank-32 but i never heard about detection of > far-end disconnection. >Does anyone have some experience about that functionality in this > hardware ? >Thanks in Advan

Re: [asterisk-users] Mark Spencer: Digium is Growing Up (VONMAG)

Maybe his comments were taken out of context as they don't have the whole interview posted. Why is he talking about queue games, Biologicall and other extremely niche crap when there are huge holes in the basic offering (SLA and SCA)? From: Al lists [mailto:[EMAIL PROTECTED] Sent: Tuesday, S

Re: [asterisk-users] Flash IDE

Quoting Gordon Henderson <[EMAIL PROTECTED]>: > On Wed, 12 Sep 2007, Juan Sandro wrote: > >> >>> You could read the archives from a week or 2 ago under the >>> heading:> Build your own "appliance" >> >> Yap... read it, thanks I use these deices, but I unload them entirely into RAM. >> >> F

Re: [asterisk-users] Flash IDE

On Wed, 12 Sep 2007, Juan Sandro wrote: > >> You could read the archives from a week or 2 ago under the heading:> Build >> your own "appliance" > > Yap... read it, thanks >>> I use these deices, but I unload them entirely into RAM. > > Fine.. I though about that too but what about: > > - if power

[asterisk-users] Astribank 32 and Far End Disconnection

Hi all, I'm interested in buy a Astribank-32 but i never heard about detection of far-end disconnection. Does anyone have some experience about that functionality in this hardware ? Thanks in Advance, Gleidson Antonio Henriques ___ Sign up

[asterisk-users] Direct dialing to correct extension from analog lines

Hi, I have a problem with people that are calling from analog lines. We have a block of numbers 12345 - 0 to -99. Most calls are transmitting the whole number including the extension. There's no problem with that. But people calling from analog lines are connected to our asterisk box as soon as

Re: [asterisk-users] Linux-HA and Asterisk

I am also using Polycom behind NAT without problems. But my asterisk box is not natted (so I have no externip setting in my asterisk). My setup is Asterisk->internet->nat->polycom601/501/430/301 (PS: Hopefully posting in plain text after adding digium.com as a text-only domain in thunderbird :

[asterisk-users] Wanted: VoIP Engineer for Warsaw !

Peoplefone AG offers Voice over IP(VoIP) services with exceptional rates. Peoplefone is a certified partner of Siemensand AVM/FRITZ!Box . Due to our rapid growth, for our new Polish

Re: [asterisk-users] TDM2400P: Power alarm error on boot

Hi all, I solved the problem changing the module. Giorgio gincantalupo wrote: > Hi, > I have an Asterisk PBX equipped with (a Sangoma PRI card and) a Digium > TDM2400P. > I got this error inside /var/log/messages:Power alarm on module > 8, resetting! > I rebooted the PBX and this time

Re: [asterisk-users] Problems with Asterisk behind a firewall

I don't have enough experiment to help you, but I can suggest you this : http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html _ Découvrez le Blog heroic Fantaisy d'Eragon! http://eragon-heroic-fantasy.spaces

Re: [asterisk-users] Different Networks

*bump* - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com - Original Message - From: "Mike Hammett" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Friday, September 07, 2007 3:25 PM Subject: Re: [asterisk-users] Different

Re: [asterisk-users] Linux-HA and Asterisk

THAT is an issue with externip= and localnet= not being correct. Mike Clark wrote: > Eric "ManxPower" Wieling wrote: >> Polycoms work just fine behind NAT. >> > Yep, we have lots of Polycoms behind NAT working fine with Asterisk > servers on *public* IPs. However, with the HA cluster, we had t

Re: [asterisk-users] Linux-HA and Asterisk

Now why would that cause a problem if it is a decent NAT router? What specific issue did you have? Obviously NAT increases the complexity of an Asterisk and phone deployment, but it does work. Dovid B wrote: > Eric, > Try 5 polycoms behind the same NAT router. Let me know when you grab a drink

Re: [asterisk-users] Flash IDE

> I've had CF units fail in service, but it's true that reliability is> > increasing, especially as they get bigger.> > I would recommend going with > the largest CF you can afford.> > -Stephen- Thanks Stephen... That makes sence now if wear leveling is used. Juan _

Re: [asterisk-users] Linux-HA and Asterisk

Quoting Jerry Jones <[EMAIL PROTECTED]>: > How about 20+ on a Qwest DSL modem hitting our server? Works great. yeah but how many have call paths open at once ? just sitting there on hook you could probably have hundreds and still be fine. and of the call paths open how many reinvited and are

[asterisk-users] Problems with Asterisk behind a firewall

Hi all, I have set up Asterisk and I am able to register with my SIP provider and receive calls. When I try to register with Asterisk from outside I can place calls but tthe other person can't hear me. Have opened port 5060 UDP as well as port 1 to 2 UDP. Any ideas? Thanks, Christian ___

Re: [asterisk-users] Flash IDE

there is tons of information about linux and flash drives on the nslu2-linux.org and the openwrt sites. main points : - disable swap - disable atime - disable most logging once the drive is not being written to then it will last a long time. Quoting Bill Seddon <[EMAIL PROTECTED]>: > Doe

Re: [asterisk-users] Linux-HA and Asterisk

How about 20+ on a Qwest DSL modem hitting our server? Works great. On Sep 12, 2007, at 7:23 AM, Dovid B wrote: > Eric, > Try 5 polycoms behind the same NAT router. Let me know when you > grab a drink > ;) > > - Original Message - > From: "Eric "ManxPower" Wieling" <[EMAIL PROTECTED]>

Re: [asterisk-users] Flash IDE

> So basically it seems that given a large enough flash drive with decent > > wear levelling the lifetime should be completely ample...> > ...Thats the > theory anyway.> > I feel quite bullish about the whole thing, but I think I > would avoid > the *really* discounted cheapo flash drives since

Re: [asterisk-users] Flash IDE

Does it have to be a flash device? I have an 8GB "flash" drive that is really a small hard disk that plugs into and is powered by a USB port. The device is 3cmx3cmx0.5cm, silent, fast and wears out like a hard disk not flash memory. It doesn't stick out (and so get knocked off) because the USB co

Re: [asterisk-users] Linux-HA and Asterisk

Eric "ManxPower" Wieling wrote: > Polycoms work just fine behind NAT. > Yep, we have lots of Polycoms behind NAT working fine with Asterisk servers on *public* IPs. However, with the HA cluster, we had the Asterisk servers NATed in a Linux-HA cluster and in that configuration, the Asterisk se

Re: [asterisk-users] Flash IDE

> You could read the archives from a week or 2 ago under the heading:> Build > your own "appliance" Yap... read it, thanks > > I use these deices, but I unload them entirely into RAM. Fine.. I though about that too but what about: - if power fails? - how/when to write changes to DOM? > If y

Re: [asterisk-users] Chan_sip Entry

Hello, Yes, I also believe that this is some sort of codec issue. Here is my sip.conf file: [201] type=friend ;secret=201 record_out=Adhoc record_in=Adhoc qualify=no port=5060 nat=no host=dynamic dtmfmode=rfc2833 dial=SIP/201 context=from-internal canreinvite=no callerid=device <

Re: [asterisk-users] SIP Debugging to separate log file

- Original Message - From: "Jason Martin" <[EMAIL PROTECTED]> To: Sent: Thursday, September 06, 2007 4:58 PM Subject: [asterisk-users] SIP Debugging to separate log file > Hello, I'm working with our SIP provider to nail down some call quality > issues > we're having, and they've aske

Re: [asterisk-users] Linux-HA and Asterisk

Eric, Try 5 polycoms behind the same NAT router. Let me know when you grab a drink ;) - Original Message - From: "Eric "ManxPower" Wieling" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Wednesday, September 12, 2007 2:43 PM Subject: Re: [asteri

[asterisk-users] fax and answer machine detection for outgoing call on DIVA card

Hello, I need to detect both fax and answer machine, and it should be valuable that the detection will be run by the Diva card itself. So : - I read Diva Documentation, and I found that the Diva could send some specific DTMF, if I had "[..] enabled [this functionnality] by the ap

Re: [asterisk-users] Linux-HA and Asterisk

Polycoms work just fine behind NAT. Mike Clark wrote: > Chris Mason (Lists) wrote: >> Mike Clark wrote: >> >> >>> Yes, the Asterisk boxes were on private addresses. The Polycoms are also >>> behind a NAT. Yes, I tried using externip in sip.conf and this allowed >>> registration, and calls to

Re: [asterisk-users] TDM400P (TDM22P) and aux power.

>>> On 9/5/2007 at 10:56 AM, Jason Parker <[EMAIL PROTECTED]> wrote: > Joe Acquisto wrote: >> I need to ask, to refresh, is the aux power connector on the TDM400P card > *only* to power the ringer on any >> analog phones/devices on the system? >> >> Can I still use this board, to "terminate"

Re: [asterisk-users] Linux-HA and Asterisk

Chris Mason (Lists) wrote: > Mike Clark wrote: > > >> Yes, the Asterisk boxes were on private addresses. The Polycoms are also >> behind a NAT. Yes, I tried using externip in sip.conf and this allowed >> registration, and calls to be placed, but no audio. Unfortunately, >> Polycom does not su

[asterisk-users] TDM2400P: Power alarm error on boot

Hi, I have an Asterisk PBX equipped with (a Sangoma PRI card and) a Digium TDM2400P. I got this error inside /var/log/messages:Power alarm on module 8, resetting! I rebooted the PBX and this time I got:Power alarm on module 7, resetting! Please, does anybody know what it mea

Re: [asterisk-users] TDM400P periodic sound clicks on FXS

Tzafrir Cohen wrote: > How about calls from either the card or the trunk to an echo test > extension? to a local SIP/IAX phone? > > It seems that the FXS slots do no have the issue with local VoIP phones. Any ideas? ___ Sign up now for AstriCon 20

[asterisk-users] TE405P intermittent yellow alarm

Folks, I really hope you can help me here - I'm beginning to tear my hair out! About 10 days ago my company moved to a new office. As a result of this, we've plugged our PBX box, which has happily been running for the last three years, into our new E1 line. Since then, I've been seeing intermitte

[asterisk-users] AAI2UUI - how?

Hi list, on my asterisk machine I have an E1 (Beronet with chan_misdn) board and sip clients connected. I am getting some AAI (application-to-application-information, enriched SIP header, similar to the SipAddHeader application) from a sip client during the BYE method. I want to give this AAI

Re: [asterisk-users] Generating an old-fashioned dialtone

Quoting Clayton Milos <[EMAIL PROTECTED]>: >> Is there a way to generate an old-fashioned dial tone with Asterisk? >> >> I'm thinking of one that sounds like: >> >> http://www.seg.co.uk/telecomm/dial tone.wav > As far as I know dialtone with SIP can only be generated on the handsets. > We're usin

Re: [asterisk-users] Generating an old-fashioned dialtone

On Wed, 12 Sep 2007, Clayton Milos wrote: > - Original Message - > From: "Phil Reynolds" <[EMAIL PROTECTED]> > To: > Sent: Wednesday, September 12, 2007 8:57 AM > Subject: [asterisk-users] Generating an old-fashioned dialtone > > >> Is there a way to generate an old-fashioned dial tone wi

Re: [asterisk-users] 56k modem configuration

Matthew Fredrickson schrieb: > Andrea Spadaccini wrote: >> Ciao Matthew, >> >>> I would be very surprised if chan_modem actually works... I don't think >>> I've *ever* seen it setup before. >> Well.. So there's no hope to make that modem work with Asterisk, right? > > Unless someone speaks otherw

Re: [asterisk-users] IAX2 NAT issues

I guess that you just need to add a rule to your simple router's config that permits udp 4569 from asterisk outbound to any IP address. Tim - Original Message - From: "Perssy Llamosas" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, Sept

Re: [asterisk-users] Linux-HA and Asterisk

Mike Clark wrote: > Yes, the Asterisk boxes were on private addresses. The Polycoms are also > behind a NAT. Yes, I tried using externip in sip.conf and this allowed > registration, and calls to be placed, but no audio. Unfortunately, > Polycom does not support STUN. Your problem is not Linux-

Re: [asterisk-users] Generating an old-fashioned dialtone

- Original Message - From: "Phil Reynolds" <[EMAIL PROTECTED]> To: Sent: Wednesday, September 12, 2007 8:57 AM Subject: [asterisk-users] Generating an old-fashioned dialtone > > Is there a way to generate an old-fashioned dial tone with Asterisk? > > I'm thinking of one that sounds like

[asterisk-users] Generating an old-fashioned dialtone

Is there a way to generate an old-fashioned dial tone with Asterisk? I'm thinking of one that sounds like: http://www.seg.co.uk/telecomm/dialtone.wav -- Phil Reynolds o mail: [EMAIL PROTECTED] |L_ \ / Web: http://www.tinsleyviaduct.com/phil/ (_)- \/ Waltham 66, Emley Moor 69, Droitwi

Re: [asterisk-users] Spawn extension (default, 1002, 2) exited non-zero on 'SIP/host-0819d0d0

hi, here is a more verbose log, obtained from enebling debug console from logger.conf Sep 12 12:33:01 DEBUG[3631]: chan_sip.c:10709 handle_request_invite: Checking SIP call limits for device Sep 12 12:33:01 DEBUG[3631]: chan_sip.c:6282 build_route: build_route: Contact hop: -- Executing Answe

  1   2   >