similar to: Few questions about Asterisk

Hello, I have few questions about Asterisk. I installed Asterisk from CVS on FreeBSD and I made cvsup 2 days ago. 1.I couldn't find Asterisk version using "asterisk -V" command. How can I to find version information? 2.I am using Wildcard X101P (FXO) and Wildcard TDM400P REV E/F (FXS)on it. I tried Asterisk CallerID feature, but unable to get it. I tried callerid signaling V23,

> 3. Supposed I have 2 fxo cards (right now I have one already) and 3 > fxs, and one of the fxo will have two phone (running pararell), is > there any way for * to: > a. It always dial the first fxo, if the fxo is busy or is being used > (have other people conversation), will * be able to switch it to > other fxo? Here's the approximiate the conditions of the phone.

Hello, I installed Asterisk from CVS on Redhat Linux 9 and working with chan_h323 module and g729/g723 free codecs too. My network connection diagram: ---------------------------------------------- X-lite/X-Pro-->Asterisk--chan_h323-->GnuGK--->AS5300-->PSTN boldsoft*CLI> show version Asterisk CVS-v1-0-03/24/06-15:27:08 built by root@boldsoft on a i686 running Linux I can make

Hello, I have been using FreeBSD-7.0STABLE with BIND-9.4.2 ( i guess, forget to check before upgrade) up to 2008-01-26 (yesterday). But after upgrade FreeBSD-7.0STABLE-->FreeBSD-7.1STABLE everything goes wrong. 1.BIND can't start anymore and giving me following message at /var/log/messages: . . . Jan 27 12:30:20 ns kernel: ad4: 152587MB <WDC WD1600AAJS-75PSA0 05.06H05> at

2015-02-27 10:25 GMT-06:00 A J Stiles <asterisk_list at earthshod.co.uk>: > O.K. So what does your existing Dial() statement in extensions.conf look > like? > apology, put the gateway was sangoma but is a openvox , all my outgoing calls out for this context: [my-mobile-out] exten => _NXXXXXXX,n,Dial(SIP/1003/${EXTEN},55,rT) exten =>

Hi everyone! I have a problem... We have received a couple of phone numbers for voip from a local voip-provider. The work fine directly with a Cisco 7960, but so far I've not been able yet to integrate them into Asterisk. I've tried: /etc/asterisk/extensions.conf ***** [ip-incoming]

Hi everyone! Situation: when I call from cell phone to a asterisk-connected phone, all works fine. When I call from the asterisk-connected phone (a Cisco 7960 SIP) to the cell, the connection gets made, but there is no audio going in either way... Asterisk reports the following: Sep 16 08:27:41 WARNING[245775]: chan_sip.c:2679 process_sdp: Insufficient information for SDP (m = '', c =

It seems the information out there on this topic was confusing enough to cause me to waste a lot of time, and that this topic is rehashed periodically. So now it's my turn.... I had a modem with an Intel 537EP chip on it and thought it might work with the zaptel drivers. Initially, it seemed the only issue would be the PCI device & vendor codes. Then, I ran into the problem that

These allow and disallow work with NuFone for me disallow=all allow=ulaw allow=alaw allow=gsm Jeff Message: 11 Date: Fri, 11 Mar 2005 11:15:51 +0100 From: "Edward Banfa" <edward@radform.com> Subject: [Asterisk-Users] NuFone Configuration [problem] To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com>

Hello everyone. I hope that this isn't a sore topic, but I've spent a week now reading through everything I could dig up from the archives on this topic, and I'm not yet satisfied with any conclusions presented there. I've also spent a few days hacking on the Asterisk code, trying to understand how things work better, and I'm still not ready to give up. I well-realize

Hi all! I'm trying to have asterisk route all outgoing calls out via my VOIP provider. exten => _NXXXXXXX,1,Dial,SIP/BYEXTENSION@VOIP seems to have them to in the direct direction. However, debug shows that my asterisk doesn't identify itself correctly: Sip read: SIP/2.0 100 Trying From: "Evert"<sip:asterisk@[my IP]>;tag=as0aca53fa To: <sip:[dialled

We have now got our new PRI line (10 channels, 100 numbers) connected and everything is working except the outgoing caller ID. Whatever SIP phone I'm using, the CID that's shown is the very first number... ----- s n i p ----- [default] include => outgoing include => priin [outgoing] exten => _NXXXXX.,1,Macro(dial,08${EXTEN},${RINGTIME}) ; Local number (w/o areacode) -

I am trying to place an analog outbound call from a Sipura SPA-841 through a * server with a TDM400P and 4 FXO's. When I call in from an analog line everything works fine, I can talk over the SIP phone. When I call out, * says: == Spawn extension (from-sip, [phonenumber], 1) exited non-zero on 'SIP/sipphone-d29d' -- Executing Dial("SIP/sipphone-9eb0",

Dear all I've got call queuing working when calls originate from my local site. After testing I migrated it to calls originating from our voip provider- it should ring an extension, then queue . All works well apart from if the caller hangs up when queued: the call hangs around in the queue as a phantom until one of the extensions answers it and it is destroyed Am I doing something wrong?

Hi again, Kind of sheepish about asking for help, as I have only spent a day banging my head off this... I got my new Welltech 3701a, 1FXS,1FXO gateway. I flashed it with what is seemingly the appropriate firmware (SIP V1.04). This seems to have gone ok, and it is now registering both ports ok with asterisk. For 1 minute I thought I was home free and and everything was just going to work

The REV I card shows up in the PCI table as: 02:05.0 Network controller: Tiger Jet Network Inc. Intel 537 (or 02:05.0 Class 0280: e159:0001) Subsystem: Unknown device b119:0001 But the REV E/F shows up as: 02:0d.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface (or 02:0d.0 Class 0780: e159:0001) Subsystem: Unknown device b100:0003 One

Hi, I'm curious what chip Digium is using in the latest TDM400 cards. Specifically, to my recollection, they used to use the TigerJet 320G, however somewhat recently, Tigerjet was bought out, and now the 320G is no longer produced. Maybe a better question is: is there a way I can take the latest DAHDI source and get a list of supported chipsets from it? Thanks. MCH -------------- next part

2015-02-26 10:45 GMT-06:00 A J Stiles <asterisk_list at earthshod.co.uk>: > > You just need to use call groups. > > In your chan_extra.conf (if it's an OpenVox) or chan_dahdi.conf, add > something like > group=1 > to the definition for each span. > > Now in the [globals] section of your dialplah, have something like > MOBILE=EXTRA/r1 > for an

Excuse my ignorance here, but I am desperately trying to isolate the IRQ for my TE110P card (shown below as t1xxp) Ive gone into my bios and disabled all usb , parallel, serial and some other devices, those that I needed to keep, I have moved off of IRQ 10 and onto IRQ 5, but everytime I boot up, I get usb-uhci and ehci_hcd using IRQ 10 as well as my Digium card. Does anybody know what these are

Hello again, Here's the next big issue, I thought I'd let you munch on. We are utilizing Cisco 7960's and the following entries in our extensions.conf file: Exten => 1637,1,Dial(SIP/100) Exten => _NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _NXXXXXXXXX,2,Congestion Exten => _1NXXXXXXXXX,1,Dial(SIP/${EXTEN}@sipdemo) Exten => _1NXXXXXXXXX,2,Congestion These