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...e that is answered starts conversations. but audio is one sided as i mentioned above. 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100715/07fe72d5/attachment.htm

...r.com) becomes unreachable and this error is returned NOTICE[3898]: chan_sip.c:11489 sip_reg_timeout: -- Registration for ' abc at nasir.server.com' timed out, trying again (Attempt #38) It may be a simple problem but is driving my crazy... please help me out thanks in advance Nasir Javaid > Message: 6 > Date: Tue, 11 May 2010 13:57:23 +0500 > From: Nasir Javaid <nasirjavaidnasir at gmail.com> > Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 23 > To: asterisk-users at lists.digium.com > Message-ID: > <AANLkTim3qsPCY0sY3hanDxt...

...ng insecure=invite,port disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm nat=yes qualify=yes [server2_abc] type=peer host=192.168.0.21 context=default dtmfmode=rfc2833 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 asterisk server. scenario is like this. > > on my system i am using this peer in sip.conf...

...89035. i attached sip debug trace but it was too heavy to be posted. if you say i can try posting it. is there any way for setting rtp port in dialplan. using functions like sipAddheader etc. so that i can set rtp ports 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 <zishanov at gmail.com> Subject: Re: [asterisk-users] One way audio when dialing multiple registr...

...er ip and asterisk does not know where to send rtp stream after sending it to router. how can this issue be resolved? is there something to be done at router confiurations or sip.conf parameters. I have already played with nat/qualify/canreinvite/directrtp/externip etc parameters. regards, Nasir Javaid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100803/06e05a6e/attachment.htm

...tring everything goes ok. but when i remove register string call doesn't go as expected. I would like to know if there is any feature that i can use to call sip peer and authenticate is in dial command or some feature in sip.conf i dont wanna use register string. please help. regards, Nasir Javaid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100510/316f5320/attachment.htm

...is mostly firewall problem. > Are you 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 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 ca...

...n=> _X.,n,Hangup as you 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, 11 May 2010 20:54:30 +0500 From: Vardan <hvardan71 at gmail.com> Subject: Re: [asterisk-users] asterisk-users Digest, Vol 70, Issue 24 To: asterisk-users at lists.digium.com Message-ID: <hsbujk$qk9$1 at dough.gmane.org> Content-Type: text/plain; charset=ISO-8859-...

...caller 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 Message: 12 > Date: Tue, 3 Aug 2010 07:21:06 -0400 > From: C F <shmaltz at gmail.com> > Subject: Re: [asterisk-users] RTP stream not passing through router > with port forwarding > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-u...

...ublic 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. but i want channel-id of called user. can anyone help what can i do. best regards, Nasir Javaid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100729/27253161/attachment.htm

...call will be out from server using [abc]'s account. i hope you understand what i mean. also i will like to know is there any way that i can include registration information in my dial string so that i have no need to write register => abc:mysecred at 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=ulaw allow=alaw allow=g729 extensions.conf [calltoserver2] ex...

...t may become reachable after one hour or one day. but all this is random. similarly it could be reachable from one client and unreachable from other client on the LAN. can anyone help me out what is going wrong. I think this could be network issue but don't know how to prove it thanks Nasir Javaid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100525/84001051/attachment.htm

...14> Contact: <sip:1334225544 at xxxxxxxxxxx:5060> Is there something to be done 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. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100722/ce4a45a2/attachment.htm

...o when dialing a registered user by his IP:port. I this case "rinstance" parameter is missing. when i dial "SIP/username" audio 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100728/11584028/attachment.htm

...k-users at lists.digium.com> > Message-ID: <201007291515.o6TFFv8t025793 at mail.debsinc.com> > Content-Type: text/plain; charset="us-ascii" > > From: asterisk-users-bounces at lists.digium.com > [mailto:asterisk-users-bounces at 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 user and asterisk are on > Local LAN. But my asterisk server is on public ip and use...

...ail 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 host 153.18.x.x". what is wrong here? thanks Nasir Javaid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100805/dd5e26d5/attachment.htm

Hello Everyone, I am facing a problem in defining the xml for domain creation using libvirt. I am using Xen hypervisor. domPtr = lib.virDomainDefineXML(conPtr, domainXML); I get the following error: "libvir: Xen Daemon error : XML error: failed to parse domain description" Any suggestions are welcome. Regards. -------------- next part -------------- An HTML attachment

Yes, DAHDI is a timing source and meetme depends on DAHDI for voice mixing. You can check details here http://www.asterisk.org/docs/asterisk/trunk/applications/meetme >Install DAHDI then !!? >On Thu, Dec 8, 2011 at 12:46 PM, Durgesh Mishra < >durgesh.mishra at rancoretech.com> wrote: >> In make menuselect =>application=>XXX app_meetme . I am doing confrence >>

...G4bK36df65b5;rport From: "pepsi coke" <sip:12345678901 at 79.80.x.x:5678>;tag=as42ec768c To: <sip:adf at 116.18.35.235:28614>;tag=d54e632c Call-ID: 0433af7878e3a8067a40f896382cc3a6 at 79.80.x.x CSeq: 103 BYE User-Agent: Asterisk PBX Max-Forwards: 70 Content-Length: 0 Nasir Javaid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100728/f1217587/attachment.htm

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 to "get" the channelID of > Channel B, as long as the two channels are bridged,