[asterisk-users] ilbc codec

2011-07-20 Thread michel freiha
Dear Sir, Can you confirm please if any version of asterisk does support ilbc 20ms instead of 30 ms sample frequency? Regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for

Re: [asterisk-users] Codec Conversion

2010-08-09 Thread michel freiha
Hello Miguel molina, I did what you asked, but still the voice is too bad Regards On Thu, Aug 5, 2010 at 11:38 PM, Miguel Molina wrote: > El 05/08/10 14:50, Tim Nelson escribió: > > - "michel freiha" wrote: > > > > Dear Sir, > > > > I tri

Re: [asterisk-users] Codec Conversion

2010-08-05 Thread michel freiha
Dear Sir, I tried to convert ilbc to ulaw and get the same result...Bad Voice Quality Regards On Thu, Aug 5, 2010 at 4:13 PM, Tim Nelson wrote: > - "michel freiha" wrote: > > > > Dear All, > > > > i would like to ask please if someone tried to make a

[asterisk-users] Codec Conversion

2010-08-05 Thread michel freiha
Dear All, i would like to ask please if someone tried to make a codec conversion from ilbc to g729, because i did that but the voice quality was too bad and a lot of disconnection.. Can i get your feedback regarding this issue please? regards -- _

Re: [asterisk-users] Muti Asterisk

2010-06-20 Thread michel freiha
Hello, The CPU load is great...It's 90% idle...even memory is great...60% Idle Regards On Sat, Jun 19, 2010 at 7:36 PM, Paul Belanger wrote: > On Sat, Jun 19, 2010 at 5:21 AM, michel freiha wrote: > > Waiting your reply > > > Reply: Do not cross-post to #asterisk-dev

[asterisk-users] Muti Asterisk

2010-06-19 Thread michel freiha
Dear All, I have installed 4 asterisks on the same Centos machine..>Each Asterisk has its own installation folder and use its own libraries...Everything looks great and all asterisks are doing their jobs correctly except one thing...I faced a voice quality issue...On a specific time, and after the

[asterisk-users] RTP Proxy

2009-11-05 Thread michel freiha
Hi all, I would like to ask please how to configure asterisk in order to unforce rtp traffic to pass through it and send them to a separate RTp proxy? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mai

Re: [asterisk-users] GoTo IF

2009-09-28 Thread michel freiha
valid handler regards On Mon, Sep 28, 2009 at 5:52 PM, Doug Lytle wrote: > michel freiha wrote: > > Hi all, > > > > I need a goto If statement syntax that check if a variable is not > > null then go to dialplan 1 else go to dialplan2 > > > > exten =>

[asterisk-users] GoTo IF

2009-09-28 Thread michel freiha
Hi all, I need a goto If statement syntax that check if a variable is not null then go to dialplan 1 else go to dialplan2 Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizon

[asterisk-users] AGI script

2009-09-28 Thread michel freiha
Dear All, I have a perl script running on my asterisk server...This script is running for all incoming calls...It checks if a user is registered on openSIPS server...Else it return a busy tone... I would like to ask you please about the syntax to call a dialplan defined on extensions.conf for a s

[asterisk-users] Compilation error

2009-07-17 Thread michel freiha
Dear Sir, I'm trying to install asterisk 1.6.1.1 on solaris 10...At the end of gmake I got the below error creating config.h In file included from sig.h:47, from el.h:107, from common.c:51, from editline.c:4: /usr/include/signal.h:77: error: syn

Re: [asterisk-users] Asterisk Error

2009-07-17 Thread michel freiha
ask ff00 broadcast 192.168.0.255 ether 0:3:ba:f2:d2:ea Yes I have a NIC, Up and running and I can SSH the server from that NIC Regards On Fri, Jul 17, 2009 at 3:21 PM, Steve Totaro wrote: > > > On Fri, Jul 17, 2009 at 2:08 AM, michel freiha wrote: > >> Hi all, >> &g

[asterisk-users] Asterisk Error

2009-07-16 Thread michel freiha
Hi all, Can you please let me know what the below issue mean when trying to start asterisk and how I can fix it? pbx_dundi.c: No ethernet interface found for seeding global EID You will have to set it manually. regards ___ -- Bandwidth and Colocation

[asterisk-users] Compilation error

2009-07-16 Thread michel freiha
Hi all I'm trying to install asteris 1.4.22.1 on Solaris 10...the server is V120 SUN spark...During compilation (gmake) I got the following error /vis.c -o np/vis.o_a np/vis.c: In function `svis': np/vis.c:205: error: `u_int32_t' undeclared (first use in this function) np/vis.c:205: error: (Each

Re: [asterisk-users] Force Authentication

2009-07-02 Thread michel freiha
n asterisk directly the authentication is successfully done by asterisk regards On Mon, Jun 29, 2009 at 6:06 PM, Alex Balashov wrote: > It does this by default unless you have allowguest set to yes, and/or > any insecure parameter options on any individual peers. > > -- &

[asterisk-users] Force Authentication

2009-06-29 Thread michel freiha
Hi all, i would like to ask please about how to force asterisk to ask for authentication when receiving an INVITE packet from any device? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To U

[asterisk-users] Dialplan matching problem

2009-05-19 Thread michel freiha
Hi all, I'm using asterisk in Realtime mode and all my dialplans are defined in extensions table...My problem is the following... If I have for a specific context an extension with value _X. and another entry for the same context with extension 123456, if an incoming call from the same context com

Re: [asterisk-users] Goto not matching

2009-05-15 Thread michel freiha
ook like? > > > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha > *Sent:* Thursday, May 14, 2009 14:13 > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-use

[asterisk-users] Goto not matching

2009-05-14 Thread michel freiha
Hi all, I'm using asterisk in realtime...I have a specific scenario to jump from context to another context...The call will come from a gateway registered under Test context and this call should be sent to the On-net extension as Listed in the paste bin below: http://pastebin.com/d50b2ba42 The i

[asterisk-users] run dialplan when open line

2009-04-21 Thread michel freiha
Hi all, Does asterisk support the following scenario? I need when a customer who own an endpoint registered on asterisk open the line, the asterisk will run a specific AGI script inside the endpoint context? Regards ___ -- Bandwidth and Colocation Provid

Re: [asterisk-users] Perl AGI

2009-04-08 Thread michel freiha
.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha > *Sent:* Wednesday, April 08, 2009 4:28 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] Perl AGI > > > > Hi all, > > I have the belo

[asterisk-users] Perl AGI

2009-04-08 Thread michel freiha
Hi all, I have the below peace of my AGI script...the problem here is that I cannot fetch the extension value to inside the script and assign it to another variable...I highlighted it in red #!/usr/bin/perl #use DBD::mysql; use DBI; use DBD::mysql; use Asterisk::AGI;

[asterisk-users] T38 FAX

2009-03-20 Thread michel freiha
Dear All, I'm trying to send FAX to an endpoint Behind NAT...The scenario i the following: PSTN_GW-->Asterisk-->asterisk-->OpenSIPS-->Endpoint behind NAT.. The FAX is failed and I got the following error log on asterisk: Mar 20 09:21:09] WARNING[20388]: chan_sip.c:12409 handle_response_invite:

[asterisk-users] Max concurrent calls

2009-03-20 Thread michel freiha
Hi all, I mentioned in asterisk.conf there is a property "maxcalls"...I know that this is the max number of concurrent calls but i need to know please if this entry is commented out, what is the default number of MAX concurrent calls supported by asterisk? Regards _

[asterisk-users] T.38 Problem

2009-03-05 Thread michel freiha
Hi all, I'm getting the following error when receiving a FAX on Asterisk... ERROR[11320]: chan_sip.c:12441 handle_response_invite: Got error on T.38 re-invite. Bad configuration. Peer needs to have T.38 disabled. Any suggestions? Regards ___ -- Bandwi

[asterisk-users] Predefined viables

2009-03-03 Thread michel freiha
Hi all, Does anyone knows how to add a new variable to the predefined variables sent by asterisk to AGI script? regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update opti

[asterisk-users] Unable to create channel

2009-03-03 Thread michel freiha
Dear All, I have the following warnings on my log file: [Mar 3 15:26:23] NOTICE[22639] chan_local.c: No such extension/context 0.0@default creating local channel [Mar 3 15:26:23] WARNING[22639] app_dial.c: Unable to create channel of type 'LOCAL' (cause 0 - Unknown) [Mar 3 15:26:24] WARNING

[asterisk-users] Asterisk Dial plan issue

2009-03-02 Thread michel freiha
Hi all, I'm using asterisk in real time mode...My extensions.conf table contains: [default] switch => Realtime/defa...@extensions I have added the following to extensions.conf table; context:micho exten: _X. priority: 1 app:Dial appdata: SIP/00xxx...@pstn GAteway Asterisk server is connected s

Re: [asterisk-users] Asterisk realtime

2009-03-02 Thread michel freiha
Dear Sir, The issue has been solved rtcachefriends=no \and everything will work Thanks On Mon, Mar 2, 2009 at 10:31 PM, michel freiha wrote: > Hi all, > > I'm using asterisk in real time mode...All extensions are defined in table > sip_buddies...Everything looks fine and as

[asterisk-users] Asterisk realtime

2009-03-02 Thread michel freiha
Hi all, I'm using asterisk in real time mode...All extensions are defined in table sip_buddies...Everything looks fine and asterisk is reading extensions info from the sip_buddies table...The problem occurs as soon as any information on an extension is changed from sip_buddies table...Which mean,

Re: [asterisk-users] Help T.38

2009-03-01 Thread michel freiha
Dear Keven, I have just post a new email with the same body due to a member advice Regards On Sun, Mar 1, 2009 at 8:21 PM, Kevin P. Fleming wrote: > michel freiha wrote: > > Dear All, > > I have created an inbound context in sip.conf that forward incoming call > > to

[asterisk-users] Help T.38

2009-03-01 Thread michel freiha
Dear All, I have created an inbound context in sip.conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes under General context...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contai

Re: [asterisk-users] No rtp activity

2009-03-01 Thread michel freiha
face such issue Regards On Sat, Feb 28, 2009 at 7:21 PM, michel freiha wrote: > Hi all > I'm using asterisk for making PSTN calls from extensions registered on > OpenSIPS...In peak hours ,number of calls Increase dramatically to a non > logic number..When checking the calls

[asterisk-users] No rtp activity

2009-02-28 Thread michel freiha
Hi all I'm using asterisk for making PSTN calls from extensions registered on OpenSIPS...In peak hours ,number of calls Increase dramatically to a non logic number..When checking the calls using asterisk CLI I saw a lot of calls in ringing status and after 300s(rtphold timeout), asterisk releas

Re: [asterisk-users] incoming call problem

2009-02-27 Thread michel freiha
Dear David, Please find on http://pastebin.com/m69b8559d my sip.conf file Thanks a lot On Fri, Feb 27, 2009 at 1:05 PM, David fire wrote: > paste your sip.conf. > David > > 2009/2/26 michel freiha > >> Dear All, >> I have created an inbound context in SIP .conf

[asterisk-users] incoming call problem

2009-02-26 Thread michel freiha
Dear All, I have created an inbound context in SIP .conf that forward incoming call to opensips server...The problem appears as soon as I enable t38pt_udptl = yes...The Asterisk negotiate the SIP session with OpenSIPS without adding voice codec to INVITE packet...It just contains T.38 protocol...Wh

[asterisk-users] Incoming call

2009-02-24 Thread michel freiha
Dera All, I have the following scenario, A customer dial a DID number...The call is routed to a PSTN GW that send the call to asterisk... On asterisk I created an AGI Script that send the call to an extension registered on OpenSIPS server... The extension is ringing successfully, but as soon as I

Re: [asterisk-users] AGI script

2009-02-23 Thread michel freiha
Dear All, I would really need to thank you all for the great help that I got here from all of you and specially Mr. Yawar hadi for his great assist and professionalism Thanks On Mon, Feb 23, 2009 at 6:48 PM, Steve Edwards wrote: > On Mon, 23 Feb 2009, Yawar Hadi wrote: > > > so if u want to rea

Re: [asterisk-users] AGI script

2009-02-23 Thread michel freiha
to read extension then supplu variable name like > $myno=$AGI->get_variable('EXTEN'); > hope u get it > > > > On Mon, Feb 23, 2009 at 1:14 PM, michel freiha wrote: > >> Dear Sir, >> >> Kindly note that the problem is on command $AGI->get_variable(

Re: [asterisk-users] AGI script

2009-02-23 Thread michel freiha
t; > > On Mon, Feb 23, 2009 at 12:23 PM, Yawar Hadi wrote: > >> How ? >> >> >> On Mon, Feb 23, 2009 at 12:09 PM, Steve Edwards > sedwards.com> wrote: >> >>> On Mon, 23 Feb 2009, Yawar Hadi wrote: >>> >>> > dear s

Re: [asterisk-users] AGI script

2009-02-22 Thread michel freiha
>> >> Luis Morales. >> >> On Fri, Feb 20, 2009 at 7:24 PM, michel freiha wrote: >> > Dear Sir, >> > >> > I need the followingA customer will dial a specific number like >> 112,this >> > will fire a php AGI script...The AGI script wi

[asterisk-users] Intel Vs AMD

2009-02-22 Thread michel freiha
Hi all, I took my decision to use Asterisk server for handling my VOIP calls...My next step is to choose the best hardware that I should use i order to have the best performance...Here I faced 2 choices for my hardware (CPU)... 1- Using Intel CPU or AMD 2- Use 32 or 64 bits Can you help me please

Re: [asterisk-users] Network architecture

2009-02-22 Thread michel freiha
Hi all, Here I a have a question and hope that someone give me the right answer...Is it better to use Intel CPU inside the hardware where I need to install Asterisk or AMD? It's better to use 32 bits or 64 bit and what is the difference between both of them? Thanks a lot On Thu, Feb 19, 2009 at

Re: [asterisk-users] DIAL() application 'g' option

2009-02-21 Thread michel freiha
hi all, I have the script but i have a small issue and need some help..can you help plz? On 2/21/09, Geoff Lane wrote: > Hi All, > > Asterisk 1.4.12 on CentOS 5 > > I'm trying to increment an AstDB key with the length of the last > outgoing call. Here's what I've got for "01" UK geographical numb

Re: [asterisk-users] AGI script

2009-02-21 Thread michel freiha
nly written as user instead of /user on first line let >>>> be go through the problem of not reading the vairable and reply back to you >>>> soon. >>>> wait and dont lose your interest .this is the way to learn some thing >>>> new .wait let me to c

Re: [asterisk-users] AGI script

2009-02-20 Thread michel freiha
FROM test where extension = '' regards On Sat, Feb 21, 2009 at 12:06 AM, michel freiha wrote: > Dear All, > > i found the issue...it was just the #!/user/bin/perl instead of > #!user/bin/perl > > Thanks a lot for all help > > Regards > > On Fri, Feb 20,

Re: [asterisk-users] AGI script

2009-02-20 Thread michel freiha
Dear All, i found the issue...it was just the #!/user/bin/perl instead of #!user/bin/perl Thanks a lot for all help Regards On Fri, Feb 20, 2009 at 11:58 PM, michel freiha wrote: > Dear Kai, > > If I change directory to /usr/bin and type the following command I got: > > [r

Re: [asterisk-users] AGI script

2009-02-20 Thread michel freiha
Dear Kai, If I change directory to /usr/bin and type the following command I got: [r...@switch1 bin]# ./perl /var/lib/asterisk/agi-bin/dial.pl VERBOSE "AGI Environment Dump:" 3 GET VARIABLE extension VERBOSE "my dialed no is :" VERBOSE "Query is:SELECT dst_nb FROM test where extension = ''"

Re: [asterisk-users] AGI script

2009-02-20 Thread michel freiha
? Thanks On Fri, Feb 20, 2009 at 10:23 PM, Steve Edwards wrote: > On Fri, 20 Feb 2009, michel freiha wrote: > > > The file dial.pl is executable and own by root as you can see below > > > > ls -lrt > > -rwxrwxrwx 1 root root 1865 Feb 20 21:18 dial.pl > > Usin

Re: [asterisk-users] AGI script

2009-02-20 Thread michel freiha
Fri, Feb 20, 2009 at 10:15 PM, Steve Edwards wrote: > On Fri, 20 Feb 2009, michel freiha wrote: > > > I'm facing the below problem: > > == dial.pl|112: Failed to execute '/var/lib/asterisk/agi-bin/dial.pl': > No > > such file or directory > > >

Re: [asterisk-users] AGI script

2009-02-20 Thread michel freiha
Sorry, it's use Asterisk::AGI; On Fri, Feb 20, 2009 at 10:13 PM, michel freiha wrote: > Kindly note that at line 6 I have: > > se Asterisk::AGI; > > Regards > > On Fri, Feb 20, 2009 at 9:44 PM, Danny Nicholas wrote: > >> Dial.pl is >> >>1

Re: [asterisk-users] AGI script

2009-02-20 Thread michel freiha
t; *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha > *Sent:* Friday, February 20, 2009 1:41 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] AGI script

Re: [asterisk-users] AGI script

2009-02-20 Thread michel freiha
oun...@lists.digium.com] *On Behalf Of *michel freiha > *Sent:* Friday, February 20, 2009 1:41 PM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] AGI script > > > > Dear Sir, > > I'm facing the below problem: > &g

Re: [asterisk-users] AGI script

2009-02-20 Thread michel freiha
to > which u r sending ur traffic.if u want to call sip sip of from one soft > phone to second soft phone then remoce /terminator ok... like > (SIP/${DestNo}) > > mail be back if any problem. > > > On Fri, Feb 20, 2009 at 1:07 PM, michel freiha wrote: > >> Dear Yawar

[asterisk-users] real time connection failed

2009-02-20 Thread michel freiha
Dear Sir, Kindly note that I'm getting the following error when trying to register on asterisk server: [Feb 20 17:12:32] WARNING[2865]: res_config_mysql.c:358 update_mysql: MySQL RealTime: Failed to query database. Check debug for more info. I can connect to the database using the username and p

Re: [asterisk-users] AGI script

2009-02-20 Thread michel freiha
>> On Fri, 20 Feb 2009, michel freiha wrote: >> >> > I need the followingA customer will dial a specific number like >> 112,this >> > will fire a php AGI script...The AGI script will connect to the database >> and >> > see if this number (112)

Re: [asterisk-users] AGI script

2009-02-19 Thread michel freiha
registered in that table and match the 112 then asterisk will dial out the number fetched Please let me know if you need any other information Regards On Fri, Feb 20, 2009 at 1:38 AM, Luis Morales wrote: > Sure, > > what did you need exactly ? > > > On Thu, Feb 19, 2009 at 6:57 PM, mi

[asterisk-users] AGI script

2009-02-19 Thread michel freiha
Dear All, I would like to ask please if someone has a AGI script that select a value from a database and dial this value as a destination number Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing li

Re: [asterisk-users] AGI pdf book

2009-02-18 Thread michel freiha
elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf > asterisk->support books section Asterisk: The Future of > Telephony<http://www.digium.com/elqNow/elqRedir.htm?ref=http://downloads.oreilly.com/books/9780596510480.pdf> > is greate. > David > > 2009/

[asterisk-users] AGI pdf book

2009-02-18 Thread michel freiha
Dear Sir, Can someone help me please to find a free ebook talking about AGI scripting through asterisk? Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit

Re: [asterisk-users] Network architecture

2009-02-18 Thread michel freiha
Dear Helm, Kindly confirm why you do not recommend the VMs solution and if you had bad experience for it and what did you get? Regards On Tue, Feb 17, 2009 at 9:24 PM, Wilton Helm wrote: > >You may be able to split up some of the servers into multiple VMs -- maybe > five >servers with five VMs

Re: [asterisk-users] Network architecture

2009-02-18 Thread michel freiha
erisk purely as a transit > element for billing. Just because a2billing is available does not mean > you should. Far more scalable solutions are easily available. > > -- > Sent from mobile device > > On Feb 17, 2009, at 10:19 AM, michel freiha wrote: > > > Hi all, &g

[asterisk-users] Network architecture

2009-02-17 Thread michel freiha
Hi all, I'm planning to build a VOIP solution for handling SIP calls coming from endpoints registered on a specific SIP proxy...I made some research regarding network architecture and found out that the best solution is to use OpenSips as SIP proxy for registration and local calls between register

[asterisk-users] call-limit

2009-02-15 Thread michel freiha
Hi all, I'm using Asterisk in real-time mode...i need to limit the number of outgoing concurrent call per extension...Wich mean limit the number of concurrent outgoing calls to 2 at a time...I added a call-limit field to sip_buddies table and put it as 2 for an extension...I tried to make 3 concur

Re: [asterisk-users] OPTIONS packets

2009-02-12 Thread michel freiha
Thanks Johansson, Everything is OK now and you were right Regards On Thu, Feb 12, 2009 at 12:30 PM, Johansson Olle E wrote: > > 11 feb 2009 kl. 10.43 skrev michel freiha: > > > Hi all, > > I need to register asterisk on an OpenSIPS SIP Proxy...The > > Registrati

[asterisk-users] OPTIONS packets

2009-02-11 Thread michel freiha
Hi all, I need to register asterisk on an OpenSIPS SIP Proxy...The Registration is OK but my asterisk is sending OPTIONS packets to OpenSIPS and the SIP Proxy is not replying back...The issue is the UNKNOWN username that reside in the OPTIONS packet as you can see in the captured packets as you c

Re: [asterisk-users] FAX

2009-01-29 Thread michel freiha
263. > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha > *Sent:* Thursday, January 29, 2009 2:10 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [aster

Re: [asterisk-users] FAX

2009-01-29 Thread michel freiha
Then just do a make && make install on asterisk again. > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha > *Sent:* Thursday, January 29, 2009 1:35 PM >

Re: [asterisk-users] FAX

2009-01-29 Thread michel freiha
*From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha > *Sent:* Thursday, January 29, 2009 1:35 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] FAX > > >

Re: [asterisk-users] FAX

2009-01-29 Thread michel freiha
I'm getting now the below notice: rtp.c: Unknown RTP codec 100 received from 'GW address' On Thu, Jan 29, 2009 at 9:18 PM, michel freiha wrote: > Do you mean call limit on the extension or on the outgoing gateway? Kindly > note that my outbound dialpeer has m

Re: [asterisk-users] FAX

2009-01-29 Thread michel freiha
ll-limit is important because 1 call can actually > be 2-N hops. > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha > *Sent:* Thursday, January 29, 200

Re: [asterisk-users] FAX

2009-01-29 Thread michel freiha
; asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha > *Sent:* Thursday, January 29, 2009 12:27 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-users] FAX > > > > Dear Sir, > > > > When trying

Re: [asterisk-users] FAX

2009-01-29 Thread michel freiha
] chan_sip.c: Hanging up call 3058f601-47504...@14.14.14.49 - no reply to our critical packet (see doc/sip-retransmit.txt). Regards On Thu, Jan 29, 2009 at 12:04 AM, michel freiha wrote: > Dear Danny, > > Thanks a lot for the help...I'll try and let you know > > Regards > &

Re: [asterisk-users] FAX

2009-01-28 Thread michel freiha
> may slow down the process or cause undesirable results by using/accounting > for unneeded or unwanted codecs. > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel

Re: [asterisk-users] FAX

2009-01-28 Thread michel freiha
gt; > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *michel freiha > *Sent:* Wednesday, January 28, 2009 3:09 PM > > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* Re: [asterisk-use

Re: [asterisk-users] FAX

2009-01-28 Thread michel freiha
try > adding gsm or just comment out the disallow and the 2 allows. (your > recipient is using a codec that isn't ulaw or alaw). > > > -- > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium

Re: [asterisk-users] FAX

2009-01-28 Thread michel freiha
um.com] *On Behalf Of *michel freiha > *Sent:* Wednesday, January 28, 2009 9:30 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] FAX > > > > Hi all, > > When trying to send a FAX I got the following error: > > Executi

[asterisk-users] FAX

2009-01-28 Thread michel freiha
Hi all, When trying to send a FAX I got the following error: Executing [003228949...@micho:1] Dial("SIP/028949469-08466918", "SIP/ 003228949...@80.169.210.181|60") in new stack [Jan 28 17:19:20] WARNING[10040]: chan_sip.c:3039 sip_call: No audio format found to offer. Cancelling call to 003228949

[asterisk-users] T.38

2009-01-27 Thread michel freiha
Dear All, I'm trying to send Fax using T.38 protocol but the FAX is not going through..I'm getting the following error om /var/log/messages [Jan 27 16:46:31] WARNING[25435] channel.c: No path to translate from SIP/80.169.210.181-0896bfd0(4) to SIP/028949469-b7703d40(256) [Jan 27 16:46:40] WARNING

[asterisk-users] G726 Codec

2009-01-27 Thread michel freiha
Dear Sir, I would like to ask please about how I can force asterisk to send all G726 codecs without translation... g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722 g723- ---- -- -- -- -- gsm- -22

Re: [asterisk-users] PAP2T provisioning

2009-01-21 Thread michel freiha
Dear Jeff, Please find attached the autoprovisionning file of the PAP2T...Kindly let me know if you need to know how to use it Regards On Wed, Jan 21, 2009 at 1:04 AM, Jeff LaCoursiere wrote: > > Anyone have an example XML file for the PAP2T? > > Cheers, > > j > > _

Re: [asterisk-users] G729 codec

2009-01-19 Thread michel freiha
c/cpuinfo > > 2009/1/19 michel freiha > >> Dear All, >> >> I have the following CPU info on my asterisk server: >> >> Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 >> EST 2008 i686 i686 i386 GNU/Linux >> >> I need t

[asterisk-users] G729 codec

2009-01-19 Thread michel freiha
Dear All, I have the following CPU info on my asterisk server: Linux switch1.domain.net 2.6.18-92.1.22.el5 #1 SMP Tue Dec 16 12:03:43 EST 2008 i686 i686 i386 GNU/Linux I need to install G729 on the asterisk server just to pass through and not for encoding...Which G729 package do you advice me to

Re: [asterisk-users] No Audio

2008-12-23 Thread michel freiha
Dear Sir, I used several other Softphones like Skype and they are facing the same problem...It seems that the issue is global du to an undersea cable cut Regards On Mon, Dec 22, 2008 at 9:07 PM, michel freiha wrote: > Hi all, > Sometimes when making a PC to PSTN call through asterisk, I

[asterisk-users] No Audio

2008-12-22 Thread michel freiha
Hi all, Sometimes when making a PC to PSTN call through asterisk, I got no audio in both sides...tracing by wireshark, I can find that RTP packets are hitting my PC but no audio...Can someone guess what could be that issue? Maybe it's a latency issue? Regards _

Re: [asterisk-users] Voice Packets latency

2008-12-18 Thread michel freiha
fromCustomer_IP:15934 (type 18, seq 038497, ts 3706128855, len 20) Regards On Thu, Dec 18, 2008 at 10:58 AM, michel freiha wrote: > Dear All, > > I would like to ask please if there is any buffer for RTP packets sent > through asterisk and if there is a way to change the value

[asterisk-users] Voice Packets latency

2008-12-18 Thread michel freiha
Dear All, I would like to ask please if there is any buffer for RTP packets sent through asterisk and if there is a way to change the value in order to increase it Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asteri

Re: [asterisk-users] tcpdum

2008-12-15 Thread michel freiha
ch packet has a timestamp at the > beginning measured in ten thousandths (I think?) of a second. You should > be able to see the RTP packet arrive and then leave again... just subtract > the timestamps for your added latency. > > Cheers, > > j > > On Mon, 15 Dec 2008, michel frei

Re: [asterisk-users] tcpdum

2008-12-15 Thread michel freiha
Dear Sir, I would like to ask please where I can find the results for RTCP debug on...It's saved on a file or it appears in the CDRs? Regards On Tue, Dec 16, 2008 at 12:06 AM, Mark Michelson wrote: > michel freiha wrote: > > Dear Sir, > > > > What I'm interested

Re: [asterisk-users] tcpdum

2008-12-15 Thread michel freiha
ds out packets with continually > increasing TTLs and the router that drops the packet will send back a > notification, so you can "trace" each hop... > > What is it you are trying to do or measure? > > j > > On Mon, 15 Dec 2008, michel freiha wrote: > > > Dear

Re: [asterisk-users] tcpdum

2008-12-15 Thread michel freiha
bytes in the packet. > > j > > On Mon, 15 Dec 2008, michel freiha wrote: > > > *Dear All, > > I run the below tcp dump on my asterisk server > > > > tcpdump -i eth0 -n -s0 -v udp port 5060 > > > > I got the following result > > > > 20:29:

[asterisk-users] tcpdum

2008-12-15 Thread michel freiha
*Dear All, I run the below tcp dump on my asterisk server tcpdump -i eth0 -n -s0 -v udp port 5060 I got the following result 20:29:48.596867 IP (tos 0x10, ttl 64, id 0, offset 0, flags [DF], proto 17, length: 373) SIP_PROXY_IP.5060 > Asterisk_IP.5060: UDP, length 345 What i need to know please

Re: [asterisk-users] ring back tone

2008-12-14 Thread michel freiha
ack tone all the time...Is that possible? Regards On Sat, Dec 13, 2008 at 10:40 AM, Philipp Kempgen wrote: > Andrew Joakimsen schrieb: > > On Fri, Dec 12, 2008 at 18:57, Eric ManxPower Wieling > wrote: > >> Philipp Kempgen wrote: > >>> michel freiha schrieb

[asterisk-users] ring back tone

2008-12-12 Thread michel freiha
Hi all, I would like to ask please if there is a way to play a ring back tone from asterisk when the customer try to make a call...I already added the ringing function to the context in extensions .conf and it work perfectly...But the issue that the asterisk server is stoping playing back his own

[asterisk-users] asterisk latency

2008-12-11 Thread michel freiha
Dear All, I would like to ask please if there is a way to reduce latency on asterisk or to check what is causing this latency Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE o

[asterisk-users] SIP Packets

2008-12-02 Thread michel freiha
Dear Sir, My Asterisk server is sending periodically the below SIP packets Retransmitting #4 (NAT) to 68.62.168.138:5060: OPTIONS sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP XX.XX.XX.XX:5060;branch=z9hG4bK37f337ed;rport From: "asterisk" ;tag=as078bf319 To: Contact: Call-ID: [EMAIL PROTE

[asterisk-users] RTCP too short

2008-11-28 Thread michel freiha
Dear Sir, I'm running Asterisk 1.4.21.2 on a CentOS machineWhen running asterisk -rv I can see a lot of messages about RTCP too short... -- Remote UNIX connection disconnected [Nov 28 13:33:00] WARNING[24863]: rtp.c:891 ast_rtcp_read: RTCP Read too short [Nov 28 13:33:00] WARNING[1980

[asterisk-users] DTMF issue

2008-11-20 Thread michel freiha
Hi all, Kindly note that I got the below message when sending DTMF in RFC2833 through asterisk PBX...The DTMF is not going through RTCP Read too short I'm using G729 codec and asteriks Asterisk 1.4.21.2 Regards ___ -- Bandwidth and Colocation Provided

[asterisk-users] DTMF payload

2008-11-20 Thread michel freiha
Dear All, Kindly let me know please where I can fix the payload of DTMF to 101...I'm using RFC2833 Regards ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:

[asterisk-users] Voice Mail

2008-11-20 Thread michel freiha
Dear Sir, I need to configure my Voice Mail on asterisk...I made the following configuration: * extensions.conf:* exten => _999.,1,VoiceMail(${EXTEN}) exten => _999.,2,HangUp() If the customer dial 9991234 then a prompt message should ask him to enter his voice message and this what is not happe

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