search for: clibi

Displaying 17 results from an estimated 17 matches for "clibi".

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2014 Jan 20
1
Dialing a SIP URI with an ";ext=" parameter
Hi All, In the midst of trying to pilot a deployment of Microsoft Lync (mainly for non-voice collaboration, specifically IM) and integrate it with our Asterisk (11.6.0 if it matters) deployment and a "everything in one place" tool when people are out of the office. I have everything on the voice side playing nice from the Lync side (Lync->Lync, Lync->Asterisk,
2013 Dec 28
1
Convert Asterisk Appliance (AA50) to "Open" Asterisk?
Hi All, Thanks for all of the help I've been given in the past and info I've picked up from this list over the years. I have an "official" Asterisk appliance (the AA50) running my PBX at home (we previously also had an AA50 in a satellite office-that one was recently retired and replaced with Asterisk running on commodity server hardware). Anyway - the AA50
2013 Nov 13
1
SIP Presence across two servers
Hi All, We've been running Asterisk for years in our offices but just recently replaced an Asterisk Appliance* in our smaller office with an actual server, upgraded the server in hardware in our HQ location and upgrading both ends to 11.5.0 with Gareth's patch for Cisco phones. 99.99% of our endpoints are Cisco 7961Gs. Each office is more-or-less standalone for ease of management and
2008 Nov 16
1
iPhone SIP or IAX client (without proxy)?
I checked the app store and haven't found anything promising, but I figured I'd ask here. Does anyone know of a SIP or IAX client for a non-jailbroken iPhone that will communicate directly with a machine running Asterisk? I know that there's at least one offering that seems like it's essentially a proxy (App runs on iPhone, iPhone talks to 3rd party server, 3rd party server talks
2008 Dec 04
2
Possible to get "Courtesy Tone" on attended transfer?
Hi All, Is there any way to provide the user receiving an attended transfer with a tone or other audible indication that the transfer is completed (i.e. Party A calls Party B, Party B announces the call while transferring to Party C, Party C hears tone when Party B completes the transfer so that they know that they are now talking to Party A instead of Party B)? I know this is possible when
2009 Mar 30
2
iphone, skype and asterisk ...
Hi,
2008 Nov 03
1
Call quality issue across VPN-> POTS vs SIP
Hi All, Got a strange (at least IMHO) issue that doesn't make much sense to me. Basic configuration is two sites with a site-to-site (aka router-to-router) VPN. Both sites have Cisco 7961G phones [with SIP firmware] on users' desks, and the only VoIP is internal - all of our outward telecom is on POTS or Centrex-enabled POTS lines. Site 1 has a Dell PowerEdge 1950 with Asterisk built
2008 Oct 28
2
Dropped Calls / Maximum Retries Exceeded / No Reply to Our Critical Packet
Hi All, I've looked through the archives and tried several variations in Google, and I haven't found anything on-point... So I'm hoping someone here may be able to help this relative Asterisk neophyte shed some light on an issue: I have a box running Asterisk 1.4.22 in our lab with several Cisco 7961G phones and an AEX804E card (4 FXO, hardware echo cancellation). The server and all
2008 Nov 20
1
Low RX volume and half duplex/"walkie-talkie" on AEX-804E
Hi All, I have a ticket open with Digium, but based on their previous lack of support for the Asterisk Appliance, I'm not really holding my breath - and, honestly, I'm not 100% convinced it's a Digium issue in the first place (but I don't know where else to point fingers). We have an AEX-804E (PCI Express, 4 FXO ports, Hardware Echo Cancellation) in a Dell PowerEdge 1950 with
2009 Jan 16
0
No subject
"Why Siphon doesn't allow to receive a call when it doesn't run Apple doesn't accept (for the moment) an application runs in the background= . So, when Siphon doesn't run, the SIP server of your provider doesn't know= your iPhone." --_000_EC80F07C30CE3E46B2AD6B4407BE086F0C2AAD0248cworksmailcwo_ Content-Type: text/html; charset="us-ascii"
2009 Feb 02
5
"No Reply to Our Critical Packet" SIP Calls Dropped in Voicemail
Hi All, I posted this a couple weeks ago with no response, I'm hoping that someone will see it this time around and be so kind as to offer advice for resolving this issue (or point me in the direction of a better place to ask) "Some" (but not all) calls on one of our Asterisk boxes are being dropped in Voicemail -- only in voicemail -- after about 20 seconds with the error logged
2008 Oct 28
1
XML Cisco config file
Hello guys, anybody here that can help me checking out this xml file, cause I am traying to configure some cisco 7911G phones to asterisk and I can't get it done.... thanks!!!! a paste of the file is here: http://pastebin.ca/1239083 -- http://celord.blogspot.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL:
2009 Jan 16
0
No subject
"Why Siphon doesn't allow to receive a call when it doesn't run Apple doesn't accept (for the moment) an application runs in the background= . So, when Siphon doesn't run, the SIP server of your provider doesn't know your iPhone." --0015174c3c60a73ef5046656ca27 Content-Type: text/html; charset=windows-1252 Content-Transfer-Encoding: quoted-printable
2009 Feb 05
2
hardware that can accomondate 2 TDM24
Hi guys, I'm building a server that need to host 2 digium TDM24 cards. I know any 3U server with 2 PCI-E slots would do. Since I do prefer supermicro server, but getting one configured is pretty darn hard. Any suggestions here? Cheers, Kelvin Chan | Positronics Ent. Product Development | | unit 272 604-628-9330 (direct) | 8128 128th St.
2010 Apr 14
1
Ring Two Extensions Simultaneously with different caller ID values?
Hi All, We're using Asterisk 1.4, and Cisco phones exclusively (mostly the 7961G, but a few 7911Gs and one 7912G for the time being-all running the SIP firmware image, plus a few analog extensions until the next capital funding cycle). Each user has a phone at his or her desk, but there are also a growing number of "common area" phones (hallway, kitchen, conference rooms, data
2009 Jan 16
0
No subject
1. a clause in iphone Developpers agreement that forbid applications runnin= g in background, 2. lack of sip clients. Now it seems skype is available on iphones. Has someone tried it ? Along new skype capabilities in Asterisk, could it be used to hook iphones = to Asterisk for both inbound and outbound calls ? Regards --_000_EC80F07C30CE3E46B2AD6B4407BE086F0C2AAD0242cworksmailcwo_
2009 Jan 20
0
Call Dropped in Voicemail / No Reply to Our Critical Packet w/ SIP Debug
Hi All, A long time ago I posted about an issue where calls on one of our Asterisk boxes were being dropped in Voicemail (and only in voicemail) after about 20 seconds with the error logged "[Jan 19 14:33:26] WARNING[27458]: chan_sip.c:1980 retrans_pkt: Hanging up call 001d45b6-1d490088-46ef7ed6-adbeb875 at 10.2.0.203 - no reply to our critical packet (see doc/sip-retransmit.txt).". I