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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 canreinvite=yes callerid="Nasir Qazi" <12234> accountcode=6:0:abc amaflags=default disallow=all allow=ulaw allow=alaw...

...above mentioned systems, and calling from system (192.168.0.254) like this SIP/${EXTEN}@abc sends call to the abc's context [payasyougo] and from there system1 (192.168.0.20) takes charge of dialing out the number in ${EXTEN}. but when i change system 1 (192.168.0.20) to my real server (e.g. nasir.server.com) which has abc as user configured same as on system1 (192.168.0.20), call goes to [default] instead of going to [payasyougo] context and is treated as incoming call... when we use register string calls works ok on real server too. I also tried SIP/abc:mysecret/${EXTEN} and SIP/${EXTEN}...

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: asterisk-users at lists...

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, asterisk attaches some random id with username. i am dialing sip uri using "Dia...

We tried to use the CURL fn to POST json, but it's sent as form data and there seems no support for changing the Content-Type header. We switched to invoking curl in the shell. All the documentation I could find says there is just one parameter for the url and an optional second for POST body. Is there an undocumented way to set Content-Type? -------------- next part -------------- An HTML

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

...lb.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. Dr. Nasir Ali Afsar MBBS, M.Phil. Senior Lecturer in Pharmacology, College of Medicine, Alfaisal University, Al-Takhassusi Road, P.O. Box 50927, Riyadh-11533, Saudi Arabia. Tel: +966-1-2157679. nafsar@alfaisal.edu<mailto:nafsar@alfaisal.edu> drnasirpk@yahoo.com<mailto:drnasirpk@yahoo.com> www.al...

...ay 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 security feature and not a bug. Regards On Tue, Oct 2, 2018, 13:03 Olivier <oza.4h07 at gmail.com> wrote: > @Nasir: > Thanks for replying here. > > Did you met in your deployments, the kind of stability issues Carlos > reported earlier ? > > Le sam. 29 sept. 2018 à 13:32, Nasir Iqbal <nasir at ictinnovations.com> a > écrit : > >> Hi Olivior, >> >> We have recently...

...ildir Is there anything I am missing or doing wrong ? I am sorry if this is a silly question. But I am a newbie to dovecot and did a lot of google search for this error or even additional documentation for dovecot without any success. Any suggestion or advice would be highly appreciated. Regards, Nasirudheen __________________________________ Do you Yahoo!? Yahoo! SiteBuilder - Free, easy-to-use web site design software http://sitebuilder.yahoo.com

...and one 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

...ne. the big plus is that they are easy to customize and developer can use the power of browser and web to build / offer features which are not possible with regular phones. Regarding your concern about BLF or call history, for me as a being developer it is just a matter of customization. Regards Nasir Iqbal ICTBroadcast - an Auto Dialer software for ITSP <https://www.ictbroadcast.com/how-become-internet-telephony-service-provider-itsp-using-ictbroadcast-sp-edition> SMS, Fax and Voice broadcasting & Inbound / Outbound Campaigns http://www.ictbroadcast.com/ On Thu, Sep 27, 2018 at 1:0...

...r router 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

...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 other for IVR. when your call FAX extension the codec will be G711 and when user call IVR the codec must be GSM Please help me Thanks Nasir Iqbal

...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 ------------------------------ > Message: 26 > Date: Thu, 15 Jul 2010 17:09:06 +0200 > From: Jonas Kellens <jonas.kellens at telenet.be> > Subject: Re: [asterisk-users] One way audio when dialing multiple registrations > To: Asterisk Users...

Hi All, I setup auto dial on my asterisk server. The problem is asterisk does not wait for called party to answer the call but proceed to process the extension specifed in my .call file My sample call file : hannel: local/0124787924@outbound-reminder MaxRetries: 5 RetryTime: 300 WaitTime: 40 Account: Reminder context: remindem extension: s priority: 1 Set: MSG=0135.20070601.0124787924 Set:

...arty Number) -- Processing IE 161 (cs0, Sending Complete) q931.c:3509 q931_receive: call 5377 on channel 1 enters state 6 (Call Present) Sending Receiver Ready (12) -------------------------------------------------------------------------------------------------------------------- -- Regards, Nasir.

...ster string 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

Hi, I have followed steps which were mentioned on forum and given below. Still couldn't get speex working. On test calls getting error "chan_sip.c: sip_call: No audio format found to offer." # yum install speex # yum install speex-devel # cd /usr/src/asterisk # make clean # make # service asterisk stop # make install # service asterisk start Also, it is not

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

This is a newbie question, I have a class which is not derived from ActionController or ActiveRecord but I want to use logging, I tried require but still logging does not work - This class is located in a file in "model" directory. ------------------------------------ require ''logger'' class Cart def add_product(product) logger.info("Searching for product