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Does anyone have any first-hand experience with the Zoiper Business version softphone? If so what has been your experience with it? Thanks, Greg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090408/f6d5af5a/attachment.htm

...01? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at 216.26.109.22> Via: SIP/2.0/UDP 147.135.32.221:5060;branch=z9hG4bK-BroadWorks.192.168.0.3-192.168.32.221V5060-0-138966241-538634340-1288768121281- Contact: <sip:4144038968 at 147.135.32.221:5060> Supported: 100rel Max-Forwards: 69 Content-Length: 308 Content-Type: applic...

...nt to compile samba, if you add "." to the path-Variable by: (This is for bash) "export PATH=$PATH:." Best regards, and nice work with samba, Harald Husemann System Engineer Materna GmbH Vo?kuhle 37 44171 Dormund - Germany - -----Urspr?ngliche Nachricht----- Von: Gregory Malsack [mailto:gmalsack@execpc.com] Gesendet: Montag, 26. M?rz 2001 16:01 An: samba@samba.org Betreff: lost?? Hello, Why does my sco unix odt system 5 come back and tell me that there is no cc found in $PATH after typing in ./configure? Thank you for your anticipated assistance. Gregory Malsack...

> Hello All, > > History ~ > I recently took a position with a call center. At the time they had > about 50 agents in a call queue. The queue was setup to ringall. The > agents use Eyebeam softphones. Everything is local lan, no routers, > everything connected via Cisco 3600 10/100 switches. > > Now we are up to about 150 agents, and I have kept everything pretty

Can anyone tell me what might cause callers on hold in a queue to drop into agents voicemail boxes? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071217/ae6cba5a/attachment.htm

Hello All, I have a need to connect an analog device to an asterisk server. The analog device has 4 analog lines going into it (it?s a fax solution). The fax solution answers the analog call, then listens for dtmf. The dtmf code that is played tells the fax device what email address to send the fax to. All calls on our system come into the server through a PRI. The faxes come in over a PRI,

Hello Everyone, I have an installation where the client has a Microsoft SQL database that holds all of their case information. They would like the asterisk system to require users to enter a valid case number when making an outgoing call. I?m seeing some documentation regarding people using Microsoft SQL for CDR storage, however nothing regarding validating authentication using a Microsoft SQL

Does anyone know how much space the appliance has for voicemail and/or logs? Doesn't have an embedded disk from what I can see, and only a 1G flash card? -- Barry D. Hassler President, HCST http://www.hcst.net/ 937-427-9000 -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hey All, Growing call center. Currently at about 200 call center staff, running about 1000 calls per hour. Gearing up to double that. Not too sure that a single server will support that growth. So, I'm trying to come up with ways to scale the system and still maintain a simplistic design. So I'd like to bounce some ideas around. Currently I am running on a Dell 1950, dual quad core

Does anyone know of a good VOIP dialtone provider in the northern Chicago area. My client has tried Broadvoice and Mix and is having problems with latency in the middle of the traceroute between him and the provider. Thanks, Greg No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.516 / Virus Database: 269.17.13/1209 - Release Date: 1/4/2008 12:05 PM

...asterisk/full to explain the entire call flow, so I built a web interface that connected to the cel database that they could use to find it for themselves. With the cel_pgsql mod crapping out all the time, I feel like that was a waste of my time.... Thanks! Look forward to your input... Gregory Malsack Ok now for the meat Asterisk Server Dell 1950 Dual Quad Core Xeon 2.33ghz 16gb Ram CentOS 5.8 x86_64 Asterisk 1.8.22.0 Recordings are stored on a separate server via NFS with a 1gb connection SQL Server Dell 2950 Dual Dual Core Xeon 2.66ghz 16gb Ram CentOS 5.8 x86_64 One of our queues. [Confi...

Which file in Asterisk have a setting for time zone? When asterisk record incoming call in Master.csv the time is 6hr. ahead. I'm on: Canada/Mountain zone -- Joseph

Hello; If I have load up to 220 extensions with 50 concurrent calls. Can one hardware server carry all this load? What is the hardware server required for this? Regards Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130714/b63756d2/attachment.htm>

...orey Farrell) * ASTERISK-24444 - PBX: Crash when generating extension for pattern matching hint (Reported by Leandro Dardini) * ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1 (Reported by Gregory Malsack) * ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended transfer (Reported by Beppo Mazzucato) * ASTERISK-24501 - ARI: Moving a channel between bridges followed by a hangup can cause an ARI client to not receive an expected ChannelLeftBridge event before StasisEnd (Re...

...orey Farrell) * ASTERISK-24444 - PBX: Crash when generating extension for pattern matching hint (Reported by Leandro Dardini) * ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1 (Reported by Gregory Malsack) * ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended transfer (Reported by Beppo Mazzucato) * ASTERISK-24501 - ARI: Moving a channel between bridges followed by a hangup can cause an ARI client to not receive an expected ChannelLeftBridge event before StasisEnd (Re...

...orey Farrell) * ASTERISK-24444 - PBX: Crash when generating extension for pattern matching hint (Reported by Leandro Dardini) * ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1 (Reported by Gregory Malsack) * ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended transfer (Reported by Beppo Mazzucato) * ASTERISK-24501 - ARI: Moving a channel between bridges followed by a hangup can cause an ARI client to not receive an expected ChannelLeftBridge event before StasisEnd (Re...

...orey Farrell) * ASTERISK-24444 - PBX: Crash when generating extension for pattern matching hint (Reported by Leandro Dardini) * ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1 (Reported by Gregory Malsack) * ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended transfer (Reported by Beppo Mazzucato) * ASTERISK-24501 - ARI: Moving a channel between bridges followed by a hangup can cause an ARI client to not receive an expected ChannelLeftBridge event before StasisEnd (Re...

...up can cause an ARI client to not receive an expected ChannelLeftBridge event before StasisEnd (Reported by Matt Jordan) * ASTERISK-24489 - Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1 (Reported by Gregory Malsack) * ASTERISK-24498 - Segmentation fault in res_hep_rtcp on attended transfer (Reported by Beppo Mazzucato) * ASTERISK-24281 - When bridging 2 chan_sip channels, MOH not removed from on-hold channels and bridge is never destroyed after hangup. (Reported by Stefan Engstr??m) * AST...

...ected ChannelLeftBridge event before StasisEnd (Reported by Matt Jordan) - [ASTERISK-24489 <https://issues.asterisk.org/jira/browse/ASTERISK-24489>] - Crash: Asterisk crashes when converting RTCP packet to JSON for res_hep_rtcp and report blocks are greater than 1 (Reported by Gregory Malsack) - [ASTERISK-24498 <https://issues.asterisk.org/jira/browse/ASTERISK-24498>] - Segmentation fault in res_hep_rtcp on attended transfer (Reported by Beppo Mazzucato) - [ASTERISK-24281 <https://issues.asterisk.org/jira/browse/ASTERISK-24281>] - When bridging 2 chan_sip cha...

Matt, At some point you need to consider how much is too much... I run a call center with more then 125 commissioned phone sales reps and more than 60 customer service reps. We run dual servers, fiber from one provider and 6 bonded T1's from another provider. We purchase our so trunks from a wholesale company who is a major provider to resellers. Being so, their network is extremely