similar to: asterisk-users Digest, Vol 70, Issue 24

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 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, 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. [abc] type=peer username=abc secret=mysecret host=192.168.0.20 context=default dtmfmode=rfc2833 ;restrictcid=no canreinvite=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, 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,

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)

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

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.

sorry, you r right i just checked it with registration so there were astdb entries for SIP registration. anyhow after clearing settings frm astdb i tried the same scenario you advised but no luck. I think i told that i am not using server as peer but want to use a user [abc] as peer so that when ever i use dial(SIP/${EXTEN}@abc) or dial(SIP/abc/${EXTEN}) the call will be out from server using

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 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

hi there, i have posted earlier on the list but got no satisfying answer. the problem is not big. I have asterisk server directly connected with internet (79.80.x.x) and clients are behind router. clients/users are registered with asterisk and are using sipura and xlite softphone. Now problem is that when a user calls other by dialing his IP:Port (sip uri), call is connected fine and he can

hello i was wondering what is the use of "rinstance" in SIP Headers. I noticed that this parameter is visible only when there is NAT invloved. I am experiencing one way audio 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.

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, I am having very strange situation. I have my sip peer located over the internet and I am able to connect and dial to it. the problem is that, if yesterday i was connected with it via my asterisk client and dialing to it normal way, today when i run asterisk on my client, my sip peer becomes unreachable. there is no change in settings or anything else. it may become reachable after one hour

Hi, I am trying to dial a sip user via his IP:PORT Combination. i am using XYZ as target user which is registered. Asterisk server IP: 70.118.x.x calling user IP: 117.58.x.x called user IP: 117.58.x.x:5062 First I dialed my registered user in normal way like this, Dial(SIP/XYZ,30,rtT) and during conversation audio was OK in both ways. Then I dialed the registered user via

@Olivior I agree that seting up WebRTC is hard, however when done it is smooth to use. For replication you can build RPMs with working configurations. Regarding stability, it is not being used widly, so can't say it is mature. However we have no complain so far regarding audio or connectivity. sometime we provide support for "allow media / mic" type issues, but you know it is

Hi everybody, >From asterisk CLI we can add extensions in dial-plan dynamically using "dialplan add extension" command. but how we can dynamically create a context in dialplan. is that possible? Nasir Iqbal

Dear R-project team, I have tried but could not install the R statistical package (http://cran.ms.unimelb.edu.au/ ) even after the help of my institute's IT personnel. The setup file could not be downloaded. The latest file R-2.12.2.tar.gz<http://cran.ms.unimelb.edu.au/src/base/R-2/R-2.12.2.tar.gz> does not start installation wizard. Kindly extend the technical support. Best regards.

Hi All, I am wondering that how I can setup different codec for different extensions in my dial plan. scanario will when user X (Sip) call 111 extension in default context. The Asterisk response should be in GSM codec When user X (Sip) call 222 extension in default context. the Asterisk response should be in G711 Codec Actually I want to setup an extension for FAX receiving (rx_fax) and