[asterisk-users] Dialing a SIP Peer without using register strin

help. regards, Nasir Javaid -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk

Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 23

canreinvite=yes insecure=invite,port disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm nat=yes qualify=yes Nasir Javaid wrote: Hi, I am new to this list and this is first time i m posting here. please help me out currently I am working on dialing a sip peer on an asterisk server from 2nd

Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24

help me out thanks in advance Nasir Javaid Message: 6 Date: Tue, 11 May 2010 13:57:23 +0500 From: Nasir Javaid nasirjavaidna...@gmail.com Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 23 To: asterisk-users@lists.digium.com Message-ID

Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 25

can see above *highlighted that context of abc is payasyougo.*problem is that i want the call to land in that context on nasir.server.com, which works if i use register string. but without register string call goes to default context on nasir.server.com regards, Nasir Javaid Message: 19 Date: Tue

Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 25

register string call goes to default context on nasir.server.com regards, Nasir Javaid Message: 19 Date: Tue, 11 May 2010 20:54:30 +0500 From: Vardan hvarda...@gmail.com Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24 To: asterisk-users@lists.digium.com Message-ID

Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 25

...@nasir.server.com Content-Length: 0 On Wed, May 12, 2010 at 7:26 PM, Nasir Javaid nasirjavaidna...@gmail.comwrote: here i am attaching debug trace of sip in case of sccessfull call when using register string... *CLI [May 12 19:21:14] --- SIP read from 192.168.0.254:5060 --- INVITE sip

Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 30

register = abc:mysec...@nasir.server.com:8060 regards, Nasir Javaid Look, you do again with registration. remove any registration information. Look this config, I think it can help you Server1: sip.conf [interboxserver2] type=friend host=192.168.250.112 context=callfromserver2 disallow=all allow

Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 54

be network issue but don't know how to prove it thanks Nasir Javaid -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs

[asterisk-users] One way audio when dialing multiple registrations

of XYZ likeDial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends. have any idea what is going wrong?? any help will be highly appreciated regards, Nasir Javaid -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] asterisk-users Digest, Vol 72, Issue 39

behind firewall ? You can check the audio-ports that are being used in the SDP-message by doing a /sip debug/. Maybe you do not have enough UDP-ports open for the audio ? Jonas. On 07/15/2010 04:38 PM, Nasir Javaid wrote: Hi, I am working on calling 2

[asterisk-users] One way audio when dialing multiple registrations

. But simply dialing single registration of XYZ like Dial(SIP/XYZ,30,tTog) works fine and audio is fine at both ends. have any idea what is going wrong?? any help will be highly appreciated regards, Nasir Javaid

[asterisk-users] One way audio when dialing multiple registrations

a typing mistake that may have diverted all of you. hope this clears what i am trying to do. regards, Nasir Javaid --- I am sure you can't achieve what you

Re: [asterisk-users] asterisk-users Digest, Vol 72, Issue 49

with the channels involved in conversation at runtime. sincere regards, Nasir Javaid --- Message: 2 Date: Mon, 19 Jul 2010 13:41:32 -0400 From: Zeeshan Zakaria zisha...@gmail.com Subject: Re

[asterisk-users] One way audio when dialing multiple registrations

assign this parameter when dialing via IP:PORT? 2) what else options do we have if we want to dial using IP:PORT mechanism. waiting for your kind resopnse. Nasir Javaid

[asterisk-users] SIP URI Dial has one way audio

with rinstance ?? 1) how can we assign this parameter when dialing via IP:PORT? 2) what else options do we have if we want to dial using IP:PORT mechanism. waiting for your kind response. Nasir Javaid. -- _ -- Bandwidth

[asterisk-users] Nat issue one way audio on IP dial

: 0433af7878e3a8067a40f896382cc...@79.80.x.x CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Nasir Javaid -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

[asterisk-users] what is rinstance parameter in sip header

is fine but when i dial SIP/x.x.x.x:port there is one way audion. Also please tell me what can go wrong by dialing by ip:port.?? Best regards, Nasir Javaid -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

[asterisk-users] Nat issue one way audio on IP dial

thanks Jim I will check stun server settings asap, but i have noticed 192.168.x.x is also present in the debug of successful call having both way audio. so i don't think this has to do anything with this. below is the sip debug of successful call . --- Audio is at 79.80.154.99 port 14034

[asterisk-users] How to extract channel-id of a user or peer

server. I also tried built-in variable ${CHANNEL}, but this returns the channel-id of the calling channel. but i want channel-id of called user. can anyone help what can i do. best regards, Nasir Javaid -- _ -- Bandwidth

Re: [asterisk-users] asterisk-users Digest, Vol 72, Issue 81

-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Nasir Javaid Subject: [asterisk-users] How to extract channel-id of a user or peer my question is how can i get channel-id of a user or peer. I tried using ChanIsAvail(username). this works correctly when

Re: [asterisk-users] asterisk-users Digest, Vol 72, Issue 82

thanks for your reply but i think ${BRIDGEPEER} will work only when both channels are connected. i want to get channel-id before dialing so that i can dial using that channel id. ${BRIDGEPEER} is probably a good way to do what you want.. if Channel A calls Channel B, and you want Channel A

[asterisk-users] RTP stream not passing through router with port forwarding

/directrtp/externip etc parameters. regards, Nasir Javaid -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

Re: [asterisk-users] asterisk-users Digest, Vol 73, Issue 5

can hear the called party but called party can not hear caller. and there are no re-invites issued too. i am bit new to sip and rtp stuff and don't know what is going on. how asterisk is issuing re-invites for devices behind same router and not for device behind another router? Nasir Javaid

[asterisk-users] Can ChanIsAvail return status from sip uri using router ip

) calling user with this sip uri works fine. I once tried but status returned was unknow host 153.18.x.x. what is wrong here? thanks Nasir Javaid -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New