similar to: SIP multi Bindport

Hi all, I have H.323 Gateway, and i want to make a peer to route calls to this GW. But i don't know is oh323.conf supporting to add peer type entry with all feature. Please let me know how i can add H.323 GW type peer? -------- Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 ICQ: 276994704 MSN: abdulzu@hotmail.com GoogleTalk: lateef.np@gmail.com YM!: abdul_zu Doha

Hi, I have 2 g729 licences - what I want to do is use g729 by default but if I get more than 2 calls at a time, use gsm for the others. So, I put this on all my sip providers: disallow=all allow=g729 allow=gsm However, this just seems to use gsm for everything. If I comment out the gsm line, it then uses g729. I thought it would use the codec's in the order they are allowed - is this

Hello all I'm trying to get my asterisk config to listen to multiple ports. This is since some clients have port 5060 blocked by their ISP. Does anyone know how to do this in sip.conf or if it is even supported? Thanks!

Good Day All, I am facing a serious problem since I started to use asterisk. I don?t know if it is a bug or some one already solved this. Currently I am running 120 VoIP SIP channels on my asterisk server but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI ?show channels? showing us as call UP but in real there is no call. When asterisk restarted the hanged calls removed from

Hello my isp has blocked outgoing and incoming connection for port 5060 . I have ssh access to server so i want to send all traffic from port 5091 to port 5060 of asterisk .so i tried iptables -t nat -A PREROUTING -i eth0 -p udp --dport 5091 -j DNAT --to 127.0.0.1:5060 Now my softphone is able to register with asterisk but it isnt able to make any calls . bindport = 5091 in my sip.conf under

Hi All, We updated with Asterisk 1.4.17 but it seems unstable. 3, 4 times in one day it stop to response to the SIP Clinets so they cannot make call or register. But safe_asterisk not restarting it back because asterisk running without any response to the sip clients. When we try to do 'core show channels' using Manager it returns only Action: Command Command: show channels That time

I am using a month old svn version of asterisk 1.2 . I have set bindport=5091 for a sip peer ( type = friend) and nat=yes .. in sip show peer it shows port 5091 for peer but asterisk isnt listening on port 5091 at all . I tried both port=5091 as well as binport=5091 but asterisk does not listen on port 5091 . What am i doing wrong ? -------------- next part -------------- An HTML attachment was

Hello, we want to setup the following scenario: - each user has a softphone AND a hardphone - the softphone is started with the operating system - the hardphone is connected all the time using SIP - only ONE extension for each user Both phones should ring when the user is called. We've setup an asterisk 1.4.18 and at the moment only the last registered client rings. In Asterisk 1.2 the

Hi friends, Is their any possibility to setup our own Toll-Free Number in Asterisk using some PCI or FXO Card? I have one number from my local Telecom called 123XXXXXXXXXXXX and i would like to setup this number in my asterisk if some one called this number from his mobile or land line he should not be charged when the call will come i can route to my SIP or IAX in asterisk internally. In this

Hi all, I have *NOW beta 6 (asterisk 1.4.5) and I've configured it with a SIP trunk line. I can make outgoing calls, but I cannot receive any incoming calls. A port scan of my * server shows that port 5060 is closed. How do I open this port? In my users.conf, I have set [trunk_1] to hassip=yes and port=5060. Also, in the global SIP.conf file bindport=5060 bindaddr=0.0.0.0

Hi, No matter I try, I can not register to Voipwise with Trixbox. It is always in "unregistered" state in sip registry. Here is my last sip trunk configuration: PEER DETAILS: allow=g729 bindport=5060 disallow=alldtmfmode=rfc2833 fromdomain=sip.voipwise.com fromuser=username host=sip.voipwise.com insecure=very maxexpirey=120 pickupgroup=1 port=5060 secret=pass type=peer

I just see warning? 2016-08-29 11:30 GMT-03:00, Telium Technical Support <support at telium.ca>: > This shows that asterisk's IAX is already bound to all adapters - so that's > good. Symptomatically does your IAX stop working? Or do you just see a > warning? > > -----Original Message----- > From: asterisk-users-bounces at lists.digium.com >

Hi to everybody, My IAX is not working, When I type reload IAX it returns me: AsteriskSlave*CLI> iax2 reload == Parsing '/etc/asterisk/iax.conf': Found == Parsing '/etc/asterisk/users.conf': Found [Aug 26 10:05:04] NOTICE[18078]: chan_iax2.c:13546 set_config: Ignoring bindport on reload [Aug 26 10:05:04] NOTICE[18078]: chan_iax2.c:13610 set_config: Ignoring bindaddr on

Hi expets, I have installed Asterisk 1.4.11 on CentOS4 successfully without any error. But when i am trying to start asterisk with following cmd i am getting unknown command. [cybercall at ip-208-109-177-212 ~]$ asterisk -vvvvvvc -bash: asterisk: command not found [cybercall at ip-208-109-177-212 ~]$ I checked modules and other configuration files which are installed correctly. Please help me

Hi, i have written a patch for openSSH 5.8p2 which allows the user to set the local source port. The patch is as follows: diff -rupN openssh-5.8p2//readconf.c openssh-5.8p2-srcport//readconf.c --- openssh-5.8p2//readconf.c 2010-11-20 04:19:38.000000000 +0000 +++ openssh-5.8p2-srcport//readconf.c 2011-07-17 20:57:52.385044096 +0100 @@ -125,7 +125,7 @@ typedef enum { oGlobalKnownHostsFile2,

Hi I looked at few emails related to this subject. And still not sure how to solve the loop detect problem for my case iqbala at improvise:/etc/asterisk$ cat sip.conf [general] context=line1 [phone] type=friend context=phone1 secret=g00dpazzwerd bindport=5060 host=192.168.1.106 dtmfmode=rfc2833 [line] type=friend context=line1 secret=anothers33cret bindport=5061 host=192.168.1.106

Oh! In that case ignore it. Asterisk won't rebind the adapter if you've only changed parameters. The message is misleading -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Vitor Mazuco Sent: Monday, August 29, 2016 10:41 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re:

Hi all, i am wondering if i can make two leg cdr in mysql cdr table. 1st Leg : Registrar the ATA which registered to the asterisk and it normally logging in cdr table. 2nd Leg : The CDR of carrier for the example if i send call like exten => _x.,1,Dial(SIP/${EXTEN}@AT&TIP) I this cause i can get the accrue duration of call because currently we are facing some call missing not coming

Hello all, I'm having a quite simple configuration like: SIP provider <=> asterisk SIP <=> lan Everythings works fine but sometime I can't get incoming call. here are some of the logs from set debug 25 set verbosity 25 sip show debug and sip.conf and a part of extension.conf thanks in advance Reliably Transmitting (NAT) to 212.27.52.5:5060: OPTIONS sip:freephonie.net

Hello Duncan Is it possible that we can use LLVM optimization beside O1, O2, O3 along with dragonegg plugin? Regards Shahzad On Thu, Jun 7, 2012 at 10:59 PM, Abdul Wahid Memon <engrwahidmemon at gmail.com> wrote: > Thanks alot Chad for these quick and fine responses. > > Regards > > Abdul > > On Thu, Jun 7, 2012 at 10:57 PM, Chad Rosier <mcrosier at apple.com>