[Asterisk-Users] STREAM FILE question

Dear all I was wondering is there a way to advance/rewind in playback?(STREAM FILE) say 5 seconds somehow i don't think so but I thought I' would ask Thanks Jer ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/a

Re: [Asterisk-Users] Asterisk Prepaid

On Mon, 24 May 2004, usedcanon wrote: > I have a requirement for a setup with prepaid call credits. > > I am aware of the two applications available (been researching for the past > week), app_prepaid and app_rateengine. However neither of the two sound like > exactly what I want. However I was

Re: [Asterisk-Users] asterisk prompts?

Am 24.05.2004 um 04:36 schrieb hank: hello where can I get the asterisk prompts that are included in the sample config at? they are located in the sounds folder after checkout of the cvs and in /var/lib/asterisk/sounds/ after installing *. Mike ___ Ast

[Asterisk-Users] Aastra ADSI phone

I've received my Aastra 390 phone. I got the unlock procedure from the vendor, and the services button now shows all four entrys as "". When I give the phone "ADSIProg()" the phone displays: Asterisk PBX download refused Conflict with: The asterisk.adsi hasn't been changed: DESCRIPTION "Asteris

Re: [Asterisk-Users] PRI problem???

On Sun, 2004-05-23 at 21:52, Timothy R. McKee wrote: > My problem lies in random intermittent drops of calls. The entire PRI seems > to disappear, dropping all current established calls. I see occasional > printouts on an asterisk management console showing all 23 B channels > resetting with no

Re: [Asterisk-Users] Asterisk Prepaid

Have you tried the calling card sample AGI listed in the Wiki? I am not using it in production, but I tested it and it seemed straight forward. http://www.voip-info.org/wiki-Asterisk+tips+and+tricks On Sun, 2004-05-23 at 16:10, usedcanon wrote: > I have a requirement for a setup with prepaid cal

[Asterisk-Users] PRI problem???

I have just finished installing a new asterisk box at my work. The box is quite hefty, dual Zeon 2.8s with SCSI drives and 2Gb of memory. I have a 4 port Digium T1 card for channel bank and PRI access. I activated a PRI from a local CLEC (DMS-500 based, National protocol). This PRI is on slot 2

[Asterisk-Users] asterisk prompts?

hello where can I get the asterisk prompts that are included in the sample config at? thanks hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. "time is the fire in which we burn," Tollian Soran. "grudges aren't wor

[Asterisk-Users] setting the number of rings befor asterisk picks up?

hello how do I set the number of rings picks up on? I am using a single port fxo card and currently asterisk is answering after 1 or 2 rings and I want it answering after 4 5 or 6 rings thanks hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind,

Re: [Asterisk-Users] Serious NAT problems: can't call between lines on sipura

Please ignore my previous post (below), as it's not really relevant to your problem. I was in some kind of mindless auto-email processing mode and responded without fully reading your message. Too much spam, too little sleep. Geesh. -brian Brian Cuthie wrote: Bruce, I think this is related to y

[Asterisk-Users] Sipura SPA-3000 Beta

Hi All, I'm on of those brave souls who bought into the preproduction beta of the Sipura SPA-3000 FXS/FXO adapter. I've had the unit a few days and am exploring it's workings. I really want it mostly as a straightforward FXO adapter, to replace an X101p. Let me be clear, I'd love to support Digium

Re: [Asterisk-Users] Serious NAT problems: can't call between lines on sipura

Bruce, I think this is related to your firewall. You may want to take a look a posting I did a few weeks ago. http://lists.digium.com/pipermail/asterisk-users/2004-May/046511.html Something on this topic probably belongs in the wiki. -brian Bruce Komito wrote: I have a problem that is almost cert

[Asterisk-Users] HELP!!! How do I move voicemail files to a new machine?

I copied voicemail files to a replacement system. When vm tries to play the file * throws an error messages: Unexpected header size 16 unable to open fd on / How can I copy the VM to the new machine? Thanks! Paul Paul Mahler [EMAIL PROTECTED]

Re: [Asterisk-Users] NetJet and RAS

Thanks to everyone who has replied I'm thinking that maybe I need to take a step back and ask a more basic question - Is it possible to get a 56K data connection (from a normal PSTN line, dialled with a telephone number, causing a telephone to ring at your end) to work when dialing to an ISDN li

[Asterisk-Users] Serious NAT problems: can't call between lines on sipura

I have a problem that is almost certainly nat-related, but I can't figure out what's happening. Since moving the Sipura behind a NAT server (Linksys), I am no longer able to call between the two lines on the same Sipura. When I dial one extension from the other, it rings, but immediately after I

Re: [Asterisk-Users] RxFAX generates no tiff file

Steve, We have no frame slips, so we probably have a frame slip problem ;-) This may be plaguing us on our faxing. Gain and echo cancelation (ie: none) are all approximately correct, and yet still we cannot get reliable faxing through the POTS lines plugged into our FXO card on the Adit (whereas

Re: [Asterisk-Users] NetJet and RAS

Andrew Yager wrote: Hi, Last weekend I was planning to buy a physical PBX system, but instead I have been blown away by the fact that VoIP really works, that Asterisk is so easy to set up and use... and free! We're in Australia, so as I understand it, we aren't allowed to use the Zaptel cards.

Re: [Asterisk-Users] NetJet and RAS

On Mon, 2004-05-24 at 09:57 +1000, Andrew Yager wrote: > Thanks! That's good to know. Please excuse my ignorance - if we have > two telstra ISDN2 lines, which card should I get? A somewhat reasonably priced ISDN card that works with Asterisk and is sold in Australia is the AVM Fritz: http://w

[Asterisk-Users] Fwd: regulating voip - aca

The Australian Government must be feeling left out and want to stick their noses in... ... ACA moves to regulate VoIP Rodney Gedda , Computerworld 20/05/2004 10:07:07 Mid-2005 will herald a new era in voice over IP telecommunications when the Australian Communications Authority (ACA) introduces

RE: [Asterisk-Users] NetJet and RAS

Look at http://www.voip-info.org/tiki-index.php?page=Asterisk, it has lots of good information. http://www.voip-info.org/wiki-Asterisk+Hardware in particular covers hardware such as cards. Simon -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Andrew Yage

Re: [Asterisk-Users] RxFAX generates no tiff file

Hi Petr, For most people who are sure they have no frame slips, the problem usually turns out to be frame slips :-) If you are *really* sure you do not have frame slips, then uncomment the first line in t30.c, and rebuild and reinstall spandsp. The when you exchange a fax you should end up with

[Asterisk-Users] Zapata.conf setup for TE410P

Hi, I have a TE410P with 3 E1 being enabled, some how it crashes for 2 times lately, I suspect it might be the channel setup issue, can anyone tell me if following part in zapata.conf is correct? switchtype = euroisdn signalling = pri_cpe pridialplan=local group = 1 context = incoming channel =>

Re: [Asterisk-Users] NetJet and RAS

Thanks! That's good to know. Please excuse my ignorance - if we have two telstra ISDN2 lines, which card should I get? Thanks, Andrew On 24/05/2004, at 9:53 AM, Simon Brown wrote: I, too, am in Australia. I have used the X100P card and now have recently swapped to use the TDM400P card with one

RE: [Asterisk-Users] NetJet and RAS

I, too, am in Australia. I have used the X100P card and now have recently swapped to use the TDM400P card with one FXS and one FXO. Others in Australia are also using Asterisk with the Zaptel cards. Regards, Simon Brown -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTE

Re: [Asterisk-Users] IAX2 REACHABLE/UNREACHABLE

The asterisk qualify option in sip or iax sends a test packet to the remote host every minute and measures the time it takes for a response to come back. If this time frame is less than what is configured on the qualify statement or 2000ms if it is 'yes' than the host is flagged unreachable until p

[Asterisk-Users] NetJet and RAS

Hi, Last weekend I was planning to buy a physical PBX system, but instead I have been blown away by the fact that VoIP really works, that Asterisk is so easy to set up and use... and free! We're in Australia, so as I understand it, we aren't allowed to use the Zaptel cards. We need to set up ou

[Asterisk-Users] Asterisk Prepaid

I have a requirement for a setup with prepaid call credits. I am aware of the two applications available (been researching for the past week), app_prepaid and app_rateengine. However neither of the two sound like exactly what I want. However I was wondering that someone who has used it might be ab

[Asterisk-Users] creating a single user voice mail box on asterisk?

hello how do I go create a single boice mail box on asterisk? thanks hank - - Don't judge me because I'm blind. Judge me by what's inside. if you judge me because I am blind, then it is you who is blind. "time is the fire in which we burn," Tollian Soran. "grudges aren't worth holding--One who hold

RE: [Asterisk-Users] VoicePulse SIP

I use VoicePulse Connect, have done for about 6 months. I have no problems with audio quality relating to the fact that I use IAX2 as the connection protocol. I have had issues with QoS and codecs, but these were issues at my end. I've recently started trying iLBC instead of GSM. Michael On Sat

RE: [Asterisk-Users] VoicePulse SIP

Hfailed to add the illustrative bit about my installationI DO have an X100p in my * box. I'm not using it for anything more than a timing source since I'm not happy with it as an FXO. I've just recently sarted playing with the Sipura SPA-3000 as an FXO. Michael On Sat, 22 May 2004 20

Re: [Asterisk-Users] T100P HDLC configuration

All, Just now I tried Linux kernel 2.4.19, with old sethdlc utility... everything works, so the problem seems in the zaptel driver or HDLC implementation of Linux. I really can't think the problem is in Linux kernel sources because it has passed for 6 releases since they released new HDLC API...

RE: [Asterisk-Users] ZAPTEL not loading on FC2

Change your symlink to not point to the linux source tree, but rather point at /lib/modules/2.6.5-358/build, and just do a make linux26 Or apply this patch to your makefile... Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Taz Man Sent: Sunday, May

RE: [Asterisk-Users] extension pattern matching

I think may be able to do that with _[a-z][a-z]. But I haven't tried it, you need to use 2 to make sure you don't overwrite the system extensions. As I understand the * regex implimentation, you can't do _.[a-z]. to match any letters in dialplan anywhere, but that is what you really wanted I thin

RE: [Asterisk-Users] RxFAX generates no tiff file

You should Answer() your calls... In the 5000 exten, you could move your Answer to after the dial if you like... And the h exten hangs up if it doesn't exist so that's redundant, but not bad Sam [internalexten] exten => 5000,1,Answer() exten => 5000,2,Dial(SIP/mike,60,tr) exten => 5000,3,SetLang

[Asterisk-Users] ztdummy - how to test?

I've modified ztdummy to work under 2.6 (basically ditched all the uhci stuff and added a kernel timer instead). How do I test my changes are doing anything useful? zttest gives: --- Results after 14 passes --- Best: 99.975586 -- Worst: 99.975586 Is that good/bad/terrible...? Tony -- Te audire no

[Asterisk-Users] extension pattern matching

dear all, was hoping someone could give me instruction on the syntax of extension pattern matching for letters the proposed 'dial plan' is one where any letter in the dialled digits causes the pbx to assume we are dilaling a sip url and as such forward to the appropraite sip service provider was

Re: [Asterisk-Users] SIP with TerraCall Error

Try using the IP address below directly and not a hostname. The follow works for me. [terracall] type=friend host=64.69.76.33 username=##x## secret= fromuser=##x## fromdomain=pc.tt.xten.net nat=yes context=terracall-inbound [EMAIL PROTECTED] wrote: Dear All, I had try the new cvs v

RE: [Asterisk-Users] *** Asterisk Sunday News: Conferences on the phone and IRL - "in real life"

> -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Olle E. Johansson > Sent: Sunday, May 23, 2004 8:23 AM > To: Users Asterisk > Subject: [Asterisk-Users] *** Asterisk Sunday News: Conferences on the > phone and IRL - "in real life" > > >

Re: [Asterisk-Users] RxFAX generates no tiff file

I have same problem connected to PBX over E1 and sync and not slip I have latest version spanDSP I receiving 1/3 pages from faxis ? who is a problems-) I Steve Underwood wrote: Hi Troy, People had a lot of problems like this with earlier versions of spandsp. However, the latest version is pretty

[Asterisk-Users] Problems using Adtran 750 FXO and TE405P

I was informed that I had inadvertently sent my last posting in HTML format. I apologize for any trouble that this caused and I am re-posting in plain text. Any insight into what I am doing wrong here would be appreciated. Thanks, Patrick Hello, I am trying to get an Adtran 750 w/ 1 Quad FXO an

Re: [Asterisk-Users] T100P HDLC configuration

Christian, Opss.. than where is the problem? Is it in kernel or is in zaptel driver? Could you please let me know when has been broken and if anyone is working on the fix for it? Thank you. Christian Hoffmeyer wrote: - Original Message - From: "Vasyl Rublyov" <[EMAIL PROTECTED]> To: <[EM

Re: [Asterisk-Users] T100P HDLC configuration

Thank you Michael, I used that sethdlc which is in latest zaptel, sethdlc --version does not work, but "sethdlc hdlc0 --version" works sethdlc --version --version: unable to get interface information: No such device /sbin/sethdlc hdlc0 --version sethdlc version 1.15 Copyright (C) 2000 - 2003

Re: [Asterisk-Users] T100P HDLC configuration

Christian, I think this is a YMMV, situation... From what I have read in the Ham radio lists, hdlc is supposed to work _better_ on the <2.4.26 kernels, (they actually reccommend 2.6.5. This support has apparently been backported to 2.4.26.) but you must use the sethdlc utility 1.15. The proble

[Asterisk-Users] RE: snom reporting busy when it shouldn't

No, they are not in DND mode and there is nothing strange on the displays (sorry). nicolas Christian Stredicke wrote: > Did you check if the phone is in DND state? Is there anything strange on > the display? > > CS > >> -Original Message- >> From: [EMAIL PROTECTED] [mailto:asterisk-us

RE: [Asterisk-Users] IAX2 NAT / Registration Issue

His firewall is stateless. I've run into the same issue w/the sonic wall firewall on a client site. TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Sokol Sent: Sunday, May 23, 2004 11:06 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] IAX2 NAT

Re: [Asterisk-Users] Sip proxy registration help

On Sun, May 23, 2004 at 07:21:11AM +0100, Rob Franklin wrote: > > Hi All, > > I have just installed Asterisk and am trying to connect it to a SIP > account that I currently have with www.voiptalk.org but without any > success. Although I know that voiptalk do provide asterisk accounts I > don't

Re: [Asterisk-Users] e164.org

On Sun, 23 May 2004, Tony Hoyle wrote: > Simon Dorfman wrote: > > > I wonder if someone can help me understand this. Let's say I configure my > > asterisk box to use e164 and then I try to call a phone number in Germany. > > I'm in the U.S.A. So if the number I'm calling in Germany is registere

[Asterisk-Users] IAX2 NAT / Registration Issue

I have a client using IAX Phone at his office to connect to his Asterisk located at a data center. His IAX Phone connects through his office NAT gateway device (unfortunately I don't know the specific brand and model). He can make calls just fine. However, he seems to have issues receiving calls.

[Asterisk-Users] ZAPTEL not loading on FC2

Hello all, I've just installed the Fedora core 2 and tried to compile the asterisk and the zaptel drivers Asterisk went smooth but I had troubles with the zaptel. I did copy the .config file under the kernel source and make oldconfig and make include/asm ; make include/version.h ; make SUBDIRS=scri

RE: [Asterisk-Users] Caller ID with BT CD50

> > http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administr at > ion_guide_chapter09186a00801e0eb0.html#wp1113416 > > ETSI method (type 2). > > I know my CD50 still doesn't like this method, but then as the first > generation device it can be *very* fussy at the best of times -

Re: [Asterisk-Users] Failed to bind to 0.0.0.0:5060: Address already in use

On Sunday 23 May 2004 07:09, Stefan-Michael. Günther (in-put GbR) wrote: > Jay Milk worte: > >Why don't you configure the IP address instead of leaving it > > 0.0.0.0? While that should work, it's always a good idea to be > > specific about your bindings. > > I already tested the configuration wit

Re: [Asterisk-Users] How to share Zap channels in 2 Asterisk servers

Thanks it works - Original Message - From: "David J Carter" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, May 22, 2004 10:40 AM Subject: RE: [Asterisk-Users] How to share Zap channels in 2 Asterisk servers > Call the PBX300 using IAX2 from PBX200, make sure that the call go

[Asterisk-Users] IAX2 REACHABLE/UNREACHABLE

All,   I have an issue with IAX that I can’t comprehend.  Approximately every eight minutes my servers go unreachable.  They stay unreachable for exactly 10ms.  I have two servers running IAX and it happens on both servers simultaneously.  I have searched the archives and see similar issu

RE: [Asterisk-Users] Caller ID with BT CD50

On Sun, 23 May 2004, Karl Dyson wrote: > Oooh Now which setting would I need to check?? I have a Philips Onis > DECT system, which does CLI quite happily on the BT line, and I just > checked the ATA and its running : http://www.cisco.com/en/US/products/hw/gatecont/ps514/products_administratio

Re: [Asterisk-Users] app_queue and app_groupcount

Troy Settle wrote: I just disable call waiting on all my sip phones and on all zap interfaces. No problem. That is fine if it can be done, though I prefer to keep as much set-up info on the server as possible for easier admin. Do you know of a softphone with such an option? I've been unable to find

RE: [Asterisk-Users] Caller ID with BT CD50

> If you had CID to start with, I'd expect it to work - eg. you wouldn't get > it > from the POTS line but if a VOIP call came in and the ATA186 retransmitted > that in US format then a US handset would pick it up. But once I can get the cid working, I'm hoping I can persuade the DECT units to pic

RE: [Asterisk-Users] Caller ID with BT CD50

On Sun, 23 May 2004, gARetH baBB wrote: > On Sun, 23 May 2004, Karl Dyson wrote: > > > Of course, although my wife is happy with the Cisco 7905s that have > > sprung up around the house, she still likes the cordless DECT units we > > have, and so they're plugged into an ATA186. Problem is, th

RE: [Asterisk-Users] Caller ID with BT CD50

> > ATA186 firmware 3.0+ supports more formats. > > Certainly with 3.0 my BT DECT 3010 (rebadged Siemens) base copes fine. Oooh Now which setting would I need to check?? I have a Philips Onis DECT system, which does CLI quite happily on the BT line, and I just checked the ATA and its running

Re: [Asterisk-Users] Caller ID with BT CD50

Karl Dyson wrote: Of course, although my wife is happy with the Cisco 7905s that have sprung up around the house, she still likes the cordless DECT units we have, and so they're plugged into an ATA186. Problem is, they no longer display caller id due to the ATA186 not poking it out in BT format I g

RE: [Asterisk-Users] Caller ID with BT CD50

On Sun, 23 May 2004, Karl Dyson wrote: > Of course, although my wife is happy with the Cisco 7905s that have > sprung up around the house, she still likes the cordless DECT units we > have, and so they're plugged into an ATA186. Problem is, they no longer > display caller id due to the ATA186 n

[Asterisk-Users] *** Asterisk Sunday News: Conferences on the phone and IRL - "in real life"

Here in Sweden, it's supposed to be springtime. A wonderful time of the year, with sunny skies and wonderful weather. Almost summer. Today, it's not. It's winter all over again with rain and only 3 degrees celsius outside. Better to stay inside and write a weekly Asterisk newsletter :-) This week's

Re: [Asterisk-Users] T100P HDLC configuration

- Original Message - From: "Vasyl Rublyov" <[EMAIL PROTECTED]> To: <[EMAIL PROTECTED]> Sent: Saturday, May 22, 2004 10:02 PM Subject: Re: [Asterisk-Users] T100P HDLC configuration Just would like to add, of course if it is going to help: I am using Linux 2.4.26 on Linux, compiled from

Re: RE: [Asterisk-Users] Failed to bind to 0.0.0.0:5060: Address already in use

Jay Milk worte: >Why don't you configure the IP address instead of leaving it 0.0.0.0? >While that should work, it's always a good idea to be specific about >your bindings. > I already tested the configuration with bindaddr=192.168.0.101 which is the correct ip of the asterisk box - same result

RE: [Asterisk-Users] Caller ID with BT CD50

> I have a USR modem too but it's a brick unfortunately... Pity as it was > a nice modem... the right model too so it might have worked. > > I wonder what it would take to get the zaptel drivers to pick up CID (or > even a cheap conexxant modem or something like that) - these soft modems > are pr

Re: [Asterisk-Users] RxFAX generates no tiff file

Am 23.05.2004 um 04:33 schrieb Steve Underwood: How do you run rxfax? You problem is probably something to do with that. Your's is the first report I have had of no TIFF file whatsoever. [internalexten] exten => 5000,1,Dial(SIP/mike,60,tr) exten => 5000,2,SetLanguage(de) exten => 5000,3,Playback(

Re: [Asterisk-Users] e164.org

Tony Hoyle wrote: That helps (at least until VOIP calls start being charged by the minute). Maybe someone needs to implement a switch/case statement in extensions.conf for this kind of stuff at some point. Already exists, the examples on the website if a TEL field is hit they just drop out and

Re: [Asterisk-Users] Caller ID with BT CD50

Karl Dyson wrote: Well, I have a USR015630B, which, according to the FAQ supports (UK) CLI. It supports the at #cli command, but no matter what I try, it will not pick up the caller id. Lucky I already had it and didn't buy it soley for this purpose! My caller display unit (unfortunately a CD60 --

Re: [Asterisk-Users] e164.org

Duane wrote: If there is an IAX2 or SIP or H323 NAPTR record in DNS, this increments the dial plan by +1, if it's a TEL (i.e. the talking clock in china) it increments by +51, and increments by +101 if it fails, so unless you tell it to dial the number at +51 it won't use any TEL fields from DNS

[Asterisk-Users] SIP with TerraCall Error

Dear All, I had try the new cvs version asterisk to connect to TerraCall, but fail with the follow reply, anyone know how to solve this problem. NOTICE[1133742896]: Failed to authenticate on INVITE to '"account number" ;tag=as1d02a70e' Thank You. Cary LEUNG Administrator CARYNET Information C

Re: [Asterisk-Users] Asterisk slashdotted

Yes, BRAVO! The more read about it the merrierand the more participation. I've had this discussion with peers recently, the overpriced land-line model is dying a swift death...I don't know HOW many people I've talked to who have said "FSCK" the phone company...and these people are ones who d

RE: [Asterisk-Users] snom reporting busy when it shouldn't

Did you check if the phone is in DND state? Is there anything strange on the display? CS > -Original Message- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of nicolas > Sent: Sunday, May 23, 2004 5:43 AM > To: [EMAIL PROTECTED] > Subject: [Asterisk-Users]

Re: [Asterisk-Users] e164.org

Tony Hoyle wrote: Yes, same Duane as CAcert.org... So eg. if I've registered 3 different sip providers and an IAX provider, plus a couple of landlines what is it doing? I guess I'm missing the point somewhere. The point is as simple or as complex as you like it to be. You can configure things to

[Asterisk-Users] snom reporting busy when it shouldn't

I am using asterisk cvs. Incoming/Outgoing calls are working. Calling the phone when some other lines are in use on the phone is ok. What does not work though is when the phone is ringing, nobody else can call the phone anymore. That's what * is saying: -- Got SIP response 486 "Busy Here"

RE: [Asterisk-Users] Caller ID with BT CD50

Well, I have a USR015630B, which, according to the FAQ supports (UK) CLI. It supports the at #cli command, but no matter what I try, it will not pick up the caller id. Lucky I already had it and didn't buy it soley for this purpose! My caller display unit (unfortunately a CD60 -- which I've opened

Re: [Asterisk-Users] Asterisk firewall config

Hi. Il dom, 2004-05-23 alle 01:52, Tony Hoyle ha scritto: > Surely it depends on who's calling me - if they're using a SIP phone it'll > come in over the SIP port, and if they're using an IAX phone it'll come in > over the IAX port - ie there's this context in the default iax.conf: > > [guest]

Re: [Asterisk-Users] sip call using name in sip.conf

Randy Bush wrote: [foo] type=friend I do not beleive that will work for type=friend. If you use separate type=peer and type=user blocks in sip.conf it may work. Expecially if you also specify a port in the Dial(). Else, use the hostname (or a const). hmmm. then, how do i let it be dynamic if it

Re: [Asterisk-Users] VoicePulse SIP

asterisk only supports IAX2, SIP and TEL, it will only use IAX2 and SIP entries however so it is used to route via the Net if it cannot find a route via the Net or the link isn't working it will go to the next priority in your dialplan and do whatever you want, it doesn't re-configure y

Re: [Asterisk-Users] e164.org

configure your asterisk to use e164.org and make use of EnumLookup then try to call +352 818 595, if your call goes to [EMAIL PROTECTED] then you can call me for free over the net! Marc At 03:33 23.05.2004, you wrote: Dean Collins wrote: Tony, as per you inference that e164 are up to somethi