similar to: SIP Provider in India/Pakistan/Bengladesh

Displaying 20 results from an estimated 80 matches similar to: "SIP Provider in India/Pakistan/Bengladesh"

2005 Feb 09
6
Cisco 7960 Beating a Dead Horse
Hi all, So I have been reading through the docs available online and the different threads on this list, but I cannot seem to get this phone to work. I have configured the OS79XX.TXT and SIP/SEP*.cnf files (see attached), when I configure the phone to point to my tftp server and reboot it I get this message: Connection received from 10.6.0.224 on port 50608 [09/02 12:16:11.750] Read request
2005 Sep 13
1
SetCIDName question
Hi all, I tried to set the calleridname of an incoming call to get different incoming labels displayed for different incoming numbers. This does work for hidden number-calls so I can set the displayed CIDName on my cisco7960 from "CID withheld" to "abc CID withheld" If the incoming CID isn't hidden it works to use SetCallerID but not to change only the CIDName with
2009 Jul 28
3
CIsco 7960 + asterisk: hepl needed
Dear All, I'm trying to configure my new phone Cisco 7960 to work with asterisk. I followed http://www.asteriskguru.com/tutorials/cisco_7960_ip_phone_configuration.html and I got into the point where I can see on the the display line indication showing "55 <phone icon with x>" so it looks like the phone is not registered. The phone and the asterisk are in the same local
2006 Mar 15
3
Double-ring tone
I upgraded my Cisco7960 to SIP 8-2 from 7-4. Everything seems ok, works fine. Except that when I make an outbound call, I get a double-ring sound. I also found that if the target number is engaged, I get a ring sound and at the same time get a busy sound. If I revert back to 7-4, there is no problem. Anyone else had this, or any clues on how to fix it ? All of our other phones are still on
2003 Jul 18
5
Again Asterisk and VMWare - it works now!
Hi, I have succeed using Asterisk on VMWare on an Athlon@1GB with 128 MB allocated for the Linux virtual machine. I have connected this PBX with another one using IAX/GSM. I can call the other part and the sound is great, without any interruption. The phone used is a Cisco7960 with G.711, so still a codec conversion is in place (GSM/G.711) and Asterisk/VMWare Wkst performs very well. The problem
2004 Sep 21
3
chan_sccp/SEP<mac>.cnf.xml
HI all: I have spent a large amount of time configuring/installing phones connected to Asterisk. Halfway through the process I discovered that my Cisco7960 with 2 7914 expansions was not supported in the SIP protocol. After reverting to SCCP 6.0(4.0) I am now perplexed with the hassle of configuring SCCP to properly work with Asterisk. So far I have gotten the phone to dial and receive calls
2007 Jul 27
1
PAKHostOnline.com – Web Hosting | Pakistan | http://www.pakhostonline.com/
PAKHostOnline.com offered Pakistan No.1 Web Hosting, Free DOMAIN Registration / Transfer, 99% UP-Time, 24/7 Technical Support at http://www.pakhostonline.com/ --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to
2007 Jun 10
0
UNICOM, Video Conferencing in Pakistan http://www.unicom.net.pk/
UNICOM, Video Conferencing in Pakistan http://www.unicom.net.pk/ We at Unicom are pleased to inform you that we have expanded our network of video conferencing studios in all major cities of Pakistan including Karachi, Lahore, Peshawar, Islamabad, Rawalpindi, Quetta, Hayderabad, Nawabshah, Muzafarabad, Sialkot and now in Faisalabad and Gujrawala . All of our studio are equipped with professional
2004 Sep 03
0
I forgot to add my email please contact me offline we have around 300, 000 to 1/2 million minutes per month for India and Pakistan .. can ztdummy help trunk mode?
Hi all, did not find much info in lists about subj. I have ztdummy working properly because I can use conferences without any errors. But when I try to use trunk=yes, I get the following: Sep 2 21:20:51 WARNING[1137720112]: chan_iax2.c:6422 build_user: Unable to support trunking on user home' without zaptel timing Sep 2 21:20:51 WARNING[1137720112]: chan_iax2.c:6246 build_peer: Unable to
2008 Feb 20
1
Need to Connect offices in Dubai and Pakistan
Hello All We need to connect our client's offices located in Dubai and Pakistan. Suggest us some economical solution. -- Kashif Naeem Business Development Manager Hadi Telecom www.haditelecom.com Cell: +92 (0)345 4226006 Office: +92 (0)42 5692766 Email: kashif at haditelecom.com MSN: kashif__naeem at hotmail.com Gmail: meet.kashif at gmail.com Skype: kashif.naeem 302 Y Commercial Area,
2007 Aug 12
1
PAKHostOnline.com: Web Hosting Pakistan
PAKHostOnline.com offered Pakistan No.1 Web Hosting, Free DOMAIN Registration / Transfer, 99% UP-Time, 24/7 Technical Support at http://www.pakhostonline.com/ --~--~---------~--~----~------------~-------~--~----~ You received this message because you are subscribed to the Google Groups "Ruby on Rails: Talk" group. To post to this group, send email to
2003 Jun 27
1
defaultip= in sip.conf doesnt work?
I have several (various brand) sip devices with static IP's. I understand that asterisk will not accept a registration from these devices if the host= parameter is not set to 'dynamic' in sip.conf. I want calls to these extensions to be routable even before the device registers. I understand that is what defaultip= is supposed to do, but it doesn't work. I get a busy tone when
2005 Aug 26
2
Help Solving Asterisk Lockups
I am currently testing a new Asterisk installation. The server has a T100P connected to a PRI, and about 50 Polycom IP600 phones connected via the local network. Every couple hours, Asterisk randomly stops responding to all calls, both incoming on the PRI and calls from the SIP phones. I'm not sure how or where to start debugging it. When Asterisk stops working I can still connect to
2003 Oct 01
2
Directory for Cisco 7960
Hi *, does someone has a directory that works with the Cisco 7960 and astdb or mysql/ldap? Regards, Andreas _________________________________________________________________ Gaming galore at http://xtramsn.co.nz/gaming !
2003 Nov 05
1
7960 Directory, WAS: Anyone using * in a live production environment?
> -----Original Message----- > From: asterisk-users-admin@lists.digium.com > [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of > Shaun Ewing > Sent: Tuesday, November 04, 2003 7:15 PM > To: asterisk-users@lists.digium.com > Subject: Re: [Asterisk-Users] Anyone using * in a live > production environment? > [...] > The additional features are nice too.
2005 Jul 26
2
7960 SIP Firmware Upgrade Strange Problem
Hi, I am upgrading a Cisco 7960 phone from SIP V.5.1 to 6.0 and then will to go up to 7.5 However in my first attempt to go from V.5.1 to 6.0 this is hat happens: - The phone reboots - The phone then reads the file OS79XX.TXT from the TFP server. In the file I added the version "P0S3-06-0-00" - It starts upgrading firmware - Then I get the following message: (Upgrade Failed -
2004 Apr 29
9
Asterisk VS. Skype
This might have been talked about before, but I'm posting anyhow. I've got down to testing Asterisk yesterday, and I couldn't help but compare it with Skype (a Windoze only product, yet, but extremely efficient for some reason). Skype has almost unperceptible delay (LAN), while there is almost half a second of delay regardless of the codec on Asterisk. An even if we were to
2005 Feb 10
1
Cisco7960/SCCP Transfer Help?
I have a Cisco 7960 running 7.2 of their SCCP image; I am running Asterisk 1.0.5 and using the latest Sourceforge version of SCCP2. When I make a call (or receive one) the "Transfer" softkey does not show up - as a matter of fact only 2 softkeys show up (redial & something else), but those even are not active. On a 7960 running SIP the Transfer and other buttons do show up and are
2007 Apr 26
1
Cisco 7920 sccp
I am trying to register cisco 7920 to asterisk using sccp since to sip firmware upgrade to it ,but its ends with failed registration.Can you please send me a sample for sccp.conf configuring cisco 7902. Thanks -- SCCP: Accepted connection from 192.168.5.163 -- SCCP: Using ip 192.168.5.228 -- SCCP: Accepted connection from 192.168.5.163 -- SCCP: Using ip 192.168.5.228
2004 Sep 01
5
dtmf problem
Hello! I have asterisk updated from CVS on 31/8/2004 with sample configuration. I have just changed the sip.conf to register asterisk with sip proxy in out intranet. Then I can successfully make call to asterisk and go to demo IVR, but no response to dtmfs. I try to make call from several sip phones: Cisco7960, Ata186, Snom200. All of them send telephone-event in INVITE, but asterisk answers