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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,

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,

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

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

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]

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

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",

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) -

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

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

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

Hi To dial an outside line i have to dial 0. I want to have that when we dial local numbers, that is we are in the 08 area, I don't want to have to dial 08, how to set this up in asterisk 1.6? Regards /ralf ________________________________________________ Ralf Tr?skman, IT AdLibris AB, Odengatan 106, 113 22 Stockholm, Sweden Dir: +46-(0)707548074, vxl: +46-(0)85460 60 00, fax: +46-(0)85460

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

2015-02-27 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 =>

https://bugs.freedesktop.org/show_bug.cgi?id=105884 Bug ID: 105884 Summary: Firefox causes a crash in the nouveau driver on GTX 1060 Product: xorg Version: git Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: major Priority: medium Component:

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

Hi Folks, Is there a way to find out the mobile/landline carrier name based on the phone number? For example, who is the mobile carrier for (415)2345678 I had heard about some query but just don't remember how/what? Thanks in advance. Ritesh -------------- next part -------------- An HTML attachment was scrubbed... URL:

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 =

After searching around, I've been unable to to find any relevant info on this. Perhaps the group can help? I am seeing something strange with a new Sipura SPA-3000 (and I've noticed this also with an IAX softphone): When I dial 777, this dialplan (in extensions.conf) is run: exten => 777,1,Dial(Zap/1/2345678) exten => 777,n,Hangup The number is answered by the called

Is there a configuration difference for clone X100P cards versus "compatible"? I have a similar problem to what David Shaw posted earlier today. 0.5 installed OK, but mine just with one X100P clone. Default config files, edited zapata.conf per the FAQs so it includes the line channel => 1 without the semicolon. Any outgoing call attempt returns "all circuits are busy"