similar to: Asterisk 1.4.21.1

We i have settup it like this it giveme caller id agent id and date-time on gsm file but i want them to be in folder on every day basis datewise. exten => _1NXXNXXXXXX,2,Set(CALLFILENAME=${ACCOUNTCODE}-${EXTEN}-${TIMESTAMP}) exten => _1NXXNXXXXXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb) Any Idea ? Faisal > ------------------------------ > > Message: 16 >

Hello All, I am running mirror of a website i update my mirror using rsync but when i run rsync -avvvvvvzC --timeout=600 --delete --delete-after rsync.voip-info.org::voipmirror voip-info/ I get this error opening tcp connection to rsync.voip-info.org port 873 rsync: failed to connect to rsync.voip-info.org: Connection timed out (110) _exit_cleanup(code=10, file=clientserver.c, line=94):

Hi, Well David it worked after stopping the iptables.. Now what would be the rule for iptabels to allow rsync. I am not very good with iptables help will be appreciated Thanks again for help. > -------------------------------------------- > Faisal A. Ashraf > Web www.voip.com.pk > Your VoIP Solution Partner > -------------- next part -------------- HTML attachment scrubbed

Hi, Did any one have configured Asterisk+Asttapi+Collect Software? Collect is software for Debt collection agencies with PD I am having hard time configuring it with asterisk. I need help. Regards -------------------------------------------- Faisal A. Ashraf -------------- next part -------------- An HTML attachment was scrubbed... URL:

I was just on 70minute call (IAX2 -> Internet -> IAX2) and during that time I heard several "pops", or "clicks". Each time it happened, I saw the following message: Aug 19 15:36:36 NOTICE[1173711792]: rtp.c:429 ast_rtp_read: RTP: Received packet with bad UDP checksum Any ideas what causes these, and why they turn in to a "pop", instead of just silence, or a

Hi folks, I am having a small problem with asterisk-1.6.2.7 + app_fax on OpenBSD 4.7 -release. Everything seems to work fine. I have a macro which answers, receives the fax to a tiff, and then runs a script (mailfax) to convert that to pdf and email it. It all works perfectly except for some errors I am seeing in the console. After it hangs up I get a dozen or so messages in the cli

Hello List. We are having some problems using t.38 together with a Cisco voice router at one of our providers end. We are using the new digium asterisk fax module to generate the fax, and when we use together with our internal Audiocodes Mediant 2000 gateways, we have no issues what so ever, and the faxes go right through. When we send faxes to our other provider, who has cisco hardware

To me it's a error if I can't complete calls using the ATA configured to use the g726 codec. I just tried it usign a sipura SPA-2000 (preferred codec: G726-32) and I received NOTICES and WARNINGS, but I can't complete a call. On a zap channel: -- Executing Dial("SIP/2007-e4d8", "Zap/1/2217008") in new stack -- Called 1/2217008 -- Zap/1-1 answered

Hi, I like my recordings to go to date wise folder i mean to say that for example today is 20070409 so all recordings should go directly to that folder instead of one folder for whole month. and then next day's recordings should go to next date folder. so how can i do that my current monitor context is like... exten => _1NXXNXXXXXX,3,Monitor(gsm,/rec/asterisk/apr07/${CALLFILENAME},mb)

Hi list,* *I am not able to make video calls between two sip accounts. below is the information. please help me where I am missing the configuration.* Extensions.conf* exten => 111,1,Answer() same => n,Dial(SIP/2206,60,r) same => n,Hangup() *SIP.conf* [2218] type=friend secret=******* callerid="Virendra" <9172341457> host=dynamic ;

Fasiha Ashraf --- On Sat, 22/8/09, Fasiha Ashraf <feehapk@yahoo.co.in> wrote: From: Fasiha Ashraf <feehapk@yahoo.co.in> Subject: Re: [Xen-users] my bootlog To: "Boris Derzhavets" <bderzhavets@yahoo.com> Date: Saturday, 22 August, 2009, 11:12 AM Please check what wrong here grub.conf        title Fedora (2.6.30-rc6-tip)         root (hd0,6)         kernel

Hi All, I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router with a PRI card in it, handing off to a PBX and vise verse. Calls in and out are working fine except for DTMF from Asterisk to the 2600. DTMF from the 2600 to Asterisk is fine. Here are the Asterisk console warnings I get when I send DTMF from Asterisk to the 2600: == Forcing Marker bit, because SSRC has changed Jun

Hi Ashraf, In that case I think you may need to structure the code to first build the list and only at the end supply that to the SpatialLines function, something like test.func <- function(x) { tt <- list() for ( i in ... ) { ... tt[[i]] <- (whatever) } return(SpatialLines(tt)) } Eric On Tue, Sep 26, 2017 at 12:36 PM, Ashraf Afana <asafaneh at

On 6/26/07, JR Richardson <jmr.richardson at gmail.com> wrote: > Hi All, > > I have Asterisk 1.2.9.1 sending SIP calls to a Cisco 2620XM Router > with a PRI card in it, handing off to a PBX and vise verse. Calls in > and out are working fine except for DTMF from Asterisk to the 2600. > DTMF from the 2600 to Asterisk is fine. > > Here are the Asterisk console warnings

Hi all, I get "Unknown RTP codec 72 received" message in console when call in progress, but all seems OK. Calls is from X-Lite SIP softphone to PSTN over voicepulse connect (IAX) and to FWD echo test (SIP). But this message only with one SIP client, others (X-Lite too) not giving this message. All X-Lite settings are identical. Asterisk is last cvs version This what I see in console

Hello All, Has anyone ever seen this before. This only happens when i'm on phone call -- Zap/2-1 is ringing -- SIP/2203-c48d is ringing -- SIP/2202-f2ad is ringing -- SIP/2204-11cd is ringing -- SIP/2205-ce62 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- Zap/2-1 is ringing -- SIP/2205-ce62 answered Zap/1-1 -- Hungup 'Zap/2-1' Jan

I'm playing with codecs at the moment and have found some notices errors when x-lite/pro connects to asterisk with Speex or ILBC. Initially I was getting garbled sound, but after changing magic number for both codecs to 97 (as per http://www.voip-info.org/wiki-Asterisk%20phone%20xten%20xlite and http://bugs.digium.com/bug_view_page.php?bug_id=0000918) I was able to get normal voice. BUT,

Any one see these? Are they benign, or is some system tuning required to remove them? Can't seem to find a resolution in the archives. If you have a link, it would be appreciated. Jun 2 10:58:58 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 19 received Jun 2 10:58:59 NOTICE[163044272]: rtp.c:470 ast_rtp_read: Unknown RTP codec 72 received Jun 2 10:59:00 NOTICE[163044272]:

Hi Eric, Thanks for the help.But this will not solve the problem as it will generate a list and what I need is an object of class sp using SpatialLine function from sp package.So, I need to convert each matrix to coordinates and then to a line and then to a spatial line as figured in the code. My data structure is a list of 141 matrices.Each matrix represents coordinates of the river lines

Hello everyone! I tried to send a fax over SIP with an Asterisk Server in the middle (no Digium Cards, etc. installed, everything is SIP only, the PSTN-Gateway is external). Whenever I start sending a Fax to a PSTN destination, the Call gets answered and asterisk tries to build a native bridging: -- Attempting native bridge of SIP/sip.westend.com-082fd1b8 and SIP/xxx-3ef8 Then the following