Re: [asterisk-users] No audio after a reinvite changing codec ----> with SIP DEBUG!!

2011-07-01 Thread Larry Moore
On 28/06/2011 6:59 PM, Matteo Campana wrote: Hi Larry, I have the SIP debug taken from asterisk. In this debug: 1.2.3.4 ---> IP SIP PROXY 5.6.7.8 ---> IP UAC (Linksys SPA 962) 9.10.11.12 ---> IP ASTERISK to connect to the provider

Re: [asterisk-users] dialplan execution stops after ReceiveFax

2011-06-29 Thread Larry Moore
On 29/06/2011 5:13 PM, Ruben Rögels wrote: Hello, I have a noticed strange behavior in Asterisk 1.6.18.2 with ReceiveFax Digium FAX Driver: 1.6.2.0_1.3.0 (optimized for i686_32). I use a context [capi-in] for icoming ISDN calls: == [capi-in] ; Faxe fuer Ruben exten => 12345,1,Macro(faxin

Re: [asterisk-users] Problem with ReceiveFAX app from FFA

2011-06-20 Thread Larry Moore
On 20/06/2011 8:18 AM, Steve Underwood wrote: On 06/20/2011 03:38 AM, khalid touati wrote: Hi Guys, I solved temporarely my issue by kind of tricking Asterisk, I used the following line instead of the old: exten => h,n,System('/usr/local/ bin/fax2mail -p -f "${FAXFILENOEXT}" --cid-number ${CAL

Re: [asterisk-users] No audio after a reinvite changing codec

2011-06-17 Thread Larry Moore
On 18/06/2011 5:36 AM, Matteo Campana wrote: Inviato da iPhone Il giorno 16/giu/2011, alle ore 16:37, Eric Wieling ha scritto: We experience the same thing. The solution we use is to not change codecs in the middle of a call. I assumed it was an issue with our upstream. Hi Eric, this

Re: [asterisk-users] No audio after a reinvite changing codec

2011-06-16 Thread Larry Moore
On 15/06/2011 8:15 PM, Matteo Campana wrote: HI list, no idea?? :) There not much substance in the information provided for an assessment to be made. I would suggest you capture the network traffic between UAC (g711) & Asterisk UAS ensuring the snap length is large enough to capture the w

Re: [asterisk-users] Permanent restart after upgrade

2011-06-09 Thread Larry Moore
On 10/06/2011 5:32 AM, Hans Witvliet wrote: Hi all, I got three asterisk-machines, two of them acting as proxies. On one machine (sles11sp1) i got iritating messages about not bing able to find codec's and other stuff, so i thought it might be time for an update: Stupid! I went originally from

Re: [asterisk-users] any experience with cisco media gw with fax???

2011-04-15 Thread Larry Moore
On 16/04/2011 1:24 PM, Oguzhan Kayhan wrote: Hello, We have a sip trunk end point with cisco media gateway. VoIP works fine. But when we try to send faxes thru this trunk, we simply can not. Is there anybody experienced such problem and solved? How should i set sip.conf and udptl.conf. I alread

Re: [asterisk-users] T38 fax detection using g729

2011-04-14 Thread Larry Moore
On 14/04/2011 10:16 PM, Niccolò Belli wrote: Il 14/04/2011 14:34, Larry Moore ha scritto: allow=alaw,g729 ; alaw required for T.30 facsimile if T.38 fails to I didn't understand the point. If you enable both alaw and g729 it will simply use the preferred one: if it's g729 tone based

Re: [asterisk-users] T38 fax detection using g729

2011-04-14 Thread Larry Moore
On 14/04/2011 6:57 PM, Niccolò Belli wrote: Il 14/04/2011 12:25, Larry Moore ha scritto: I made a suggestion on how you could check this i.e. have your incoming call go directly to the fax extension, my 1.8.3.2 installation immediately negotiates a T.38 connection in this sceanrio, of course I

Re: [asterisk-users] T38 fax detection using g729

2011-04-14 Thread Larry Moore
On 14/04/2011 7:25 AM, Niccolò Belli wrote: Il 13/04/2011 19:54, Larry Moore ha scritto: That is because the remote endpoint, eutelia, will need to detect the Fax Tones and send the T.38 ReINVITE to you, they may not have T.38 enabled. Uhm... it's very unlikely. I made a suggestion o

Re: [asterisk-users] T38 fax detection using g729

2011-04-13 Thread Larry Moore
On 13/04/2011 10:14 PM, Niccolò Belli wrote: Hi, I continue the discussion from https://issues.asterisk.org/view.php?id=19103 If T.38 reinvite detection should still work, why it doesn't? If I use faxdetect = t38 it does never detect the fax, even using alaw. That is because the remote endpoi

Re: [asterisk-users] [OT] Yealink IP Phones

2011-04-12 Thread Larry Moore
On 12/04/2011 5:34 PM, gincantalupo wrote: Hi, I'm trying Yealink phones too but I cannot provide remote assistance to our customers using a text-based browser like lynx (I know I could use (t)ftp provisioning system but my boss does not like it). Any idea or work-around? Set the appropri

Re: [asterisk-users] spa8000 t38 faxing

2011-04-06 Thread Larry Moore
On 6/04/2011 4:27 AM, isr...@gmail.com wrote: Ok thanks I found the problem Your welcome, can I take it that you captured the packets, you then viewed them in Wireshark and that is how you discovered the issue? Larry. -- _

Re: [asterisk-users] how to check if the call is using t38 except in the sip packets

2011-04-04 Thread Larry Moore
On 4/04/2011 10:27 PM, Israel Gottlieb wrote: How could i check if the call is using t38 except looking at the sip debug? My suggestion is to capture the packets between the two end points using tcpdump, in your case you would be captuing SIP & RTP packtes between your SPA8000 & your ITSP. I

Re: [asterisk-users] spa8000 spa2102 t38 faxing

2011-03-27 Thread Larry Moore
On 28/03/2011 5:48 AM, Israel Gottlieb wrote: still no luck i hear it change to t38 but it just doesnt connect Do you have two fax devices at your end, even a fax-modem attached to a computer will do? You are going to need to provide more information such as your current configuration and

Re: [asterisk-users] spa8000 t38 faxing

2011-03-26 Thread Larry Moore
Perhaps this will help. I have a SPA8800 which has 4 x FXS & 4 x FXO ports. It took me some time to produce a working configuration. In Asterisk I have the following where 904 is the extension of the fax-modem and itsp is you VoIP Service Provider. sip.conf [general] . . faxdetect=no t

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