similar to: asterisk-users Digest, Vol 72, Issue 82

thanks for your reply but i did not meant that. ${CALLERID(DNID)} will return then number which i don't want. what i want is channel-id like if we have a user named "nasir", then we dial it as follows Dial(SIP/nasir) but actual channel-id that asterisk uses is something like " nasir-2b487e9". and on the asterisk cli we can check this when call is answered or hangup,

Hi, my question is how can i get channel-id of a user or peer. I tried using ChanIsAvail(username). this works correctly when user and asterisk are on Local LAN. But my asterisk server is on public ip and users are behind nat, and so this method says unknow host when used on public asterisk server. I also tried built-in variable ${CHANNEL}, but this returns the channel-id of the calling channel.

hello, Although my previous posts in this forum have not received satisfying answers, here is another question from me. my question is can i use ChanIsAvail function to get the status of a user in the format SPI/user-id if i provide user in sip uri like this ChanIsAvail(SIP/user at 153.18.x.x:5062) calling user with this sip uri works fine. I once tried but status returned was "unknow

Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the

Yes this scenario works on my 2 systems which are at LAN. I made one system as server (192.168.0.20) and registered from other system... it is fine but now there is a different scene. actually there is a registered user named abc at system1 (192.168.0.20) having context [payasyougo] which is used to do outbound calls. we want to use this user's context and account so that when we register

sorry for typo mistake in my last post. as from my orignal post two registration of the same user are as follows SIP/XYZ at 119.68.0.90:5060 SIP/XYZ at 202.16.34.10:5678 so dial command with unique-id i want to use will be Dial(SIP/XYZ at 192.168.0.20:5062-096afee8,30,rtT) Dial(SIP/XYZ at 192.168.0.12:64290-0966ab80,30,rtT) and not Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)

I am configuring a test Asterisk server (1.8.9.2) to practice setting a single codec globally, to avoid transcoding as much as possible. Since all of my recordings are in gsm format, I am trying to make the SIP clients use gsm everywhere. I am using Ekiga on Fedora 16 x86_64 for my tests. [root at elx2 asterisk]# cat /etc/asterisk/sip_general_additional.conf

Hi, I am trying to dial a registered user via his IP:Port mechanism, but problem is that the audio data is not reaching to dialed user. here is the scenario. caller and callee both are registered at asterisk server. asterisk server is on public ip so no port forwarding and natting necessary there. however caller and callee both are behind router and there is port forwarding enabled and nat=yes,

Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/XYZ at 192.168.0.20:5060 SIP/XYZ at 192.168.0.10:5678 i dial using following dial string Dial(SIP/XYZ at

Hi Vardan I did same as you told and deleted the SIP information in Astdb and restarted asterisk. but the result was same. as you said there might be mistake in sip.conf so i am pasting both servers configuration here.. 1- nasir.server.com [abc] username=abc type=friend secret=mysecret nat=yes mailbox=12234568 incominglimit=2 outgoinglimit=2 host=dynamic dtmfmode=rfc2833 context=payasyougo

Hi C F no asterisk and sip device are not behind same router. actually both are in different countries. how ever when caller and callee are behind same routers voice is just fine (both ways) and i can see re-INVITEs too. but when someone calls from another router then this issue arises. caller can hear the called party but called party can not hear caller. and there are no re-invites issued

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

yes, actually this scenario is on remote servers. like SIP/XYZ at 119.18.230.20:5060 SIP/XYZ at 202.68.0.90:5678 audio is ok when dialing without using ip & port as below SIP/XYZ but when i dial using below dialstring SIP/XYZ at 202.68.0.90:5678 or SIP/XYZ at 119.18.230.20:5060 then the problem arises hope you got the idea.. Nasir

Hello I can confirm that the variable DIALEDPEERNAME contains the information that I would expect in the variable BRIDGEPEER. But I read nowhere that DIALEDPEERNAME has replaced BRIDGEPEER as of Asterisk version 13 ?! So if this is not the intention, then yes this is probably a bug and should be reported. Kind regards. Jonas. On 18-09-16 19:58, Ludovic Gasc wrote: > Hi, > >

Hello a call goes out and is answered : [Sep 17 11:29:52] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b is making progress passing it to SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23420][C-00000051] app_dial.c: SIP/myprovider-0000010b answered SIP/mysippeer-00000108 [Sep 17 11:30:05] VERBOSE[23522][C-00000051] bridge_channel.c: Channel SIP/myprovider-0000010b joined

I'm wanting to capture the zap channel that a sip channel has connected to. I came across the ${BRIDGEPEER} variable documented on the wiki, and if I show channel SIP/<channel> when a call is connected I can see BRIDGEPEER as one of the channel variables. However ${BRIDGEPEER} is not set when I want it: I run a macro when the call has been connected. Does anyone have a hint on how

Hi, i am using the n-way-call dialplan solution found on voip-info. i have added its entry in applicationmap of features.conf file. the problem is......its not working. to activate the n-way call i dial *0 but nothing happens. i have played around with dtmf and codec settings but no success. the extensions and sip configuration is below if you want to have a look. I dont have any clue why its not

Thanks Vardan, I will like to know if this scenario can work when peer is not having fixed ip and we use host = nasir.server.com ? also I have set insecure=invite,port what if i use insecure=no thanks again. Message: 24 Date: Tue, 11 May 2010 10:52:14 +0500 From: Vardan <hvardan71 at gmail.com> Subject: Re: [asterisk-users] Dialing a SIP Peer without using register strin To:

Hi, I think this have been talked over several times but I couldn't find any answer. Sorry for asking. I want from dialplan, to transfer a callee to a context-extension-priority that would play a given fax file to callee (callee is supposed to be a fax number). I can get caller's channel id (with built-in CHANNEL variable). I found BRIDGEPEER but its value remains unset (see bellow)