[asterisk-users] Trouble with Incoming Callerid on Trixbox
I am having a strange issue with setting the incoming caller id on the latest version of TrixBoxCE. Right now I have it setup with a cross-over T1 cable to our production Asterisk (1.0.9) box and from the Trixbox we can send and receive calls just fine. The problem I am having is that if a number comes into the box without a number in the Caller id field, I am unable to set the caller id manually through the dial plain. But if a call comes in and it does have the number in the caller id field, I can manually set the caller id in the dial plan however I want. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Strange Echo
List, I have begun to experience a strange echo problem on our internal network. The problem starts when "User A" calls "User B", "User A" puts "User B" on hold. "User B" heres the on hold music. "User A" returns and "User B" has trouble echo. I am using FC1, Asterisk 1.0.9. This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TE411P problem
List, I just tried to swap out our 410 for a 411 and we started have problems with on of our T1's. Setup: Span 1 - Dedicated PRI for long distance. Span 2 - 12 channels fxs_gs outgoing local. 12 Channels em_w incoming DID's. I didn't have any problems with the PRI. The trouble was with the T1. We were unable to place any local calls, and all incoming DID's where garbled. What I mean by garbled, 7744 would come in a 44. I turned off all of the software echo cancel stuff in the zapata.conf. I am going to email Digium on Monday, but I am fishing here. This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom Soundpoint 600
List, I am having trouble with one of our IP600. Every five days or so, the phone locks up. This is the third 600 I have put in place. I am running asterisk 1.0.9. Has anyone had this problem with the IP600? This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX over HTTP
UDP Because of the way TCP likes to re-transmit VOIP packets. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Friday, July 22, 2005 3:42 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX over HTTP Eric Rees wrote: > We have been running IAX through OpenVPN with SSL for 6 months without > any trouble to Las Veags, and we are in Oklahoma. Most of the time, > IAX sounds better then the land line. Using UDP or using TCP? Might want to confirm by using tcpdump. -- Eric Wieling * BTEL Consulting * 504-210-3699 x2120 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] IAX over HTTP
We have been running IAX through OpenVPN with SSL for 6 months without any trouble to Las Veags, and we are in Oklahoma. Most of the time, IAX sounds better then the land line. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julio Arruda Sent: Friday, July 22, 2005 12:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] IAX over HTTP Just remember that TCP will try to retransmit your lost voice packets, what is not exactly of any use :-). VPNs with IPSec and others (CIPE and some UDP 'related' vpns) would not create this extra (and useless) overhead. I've used IAX over OpenVPN (with SSL as you), and it does work, to some level, but I would not do it for a living :-) Iassen Hristov wrote: > I disagree. Isn't running it over a VPN the same thing? > > I have been running with no problems: > a) a soft phone over a OpenVPN VPN (over TCP) > b) a soft phone over a MS PPTP VPN > c) a hard phone over a IPSec net-to-net VPN > > For the soft phone I've used X-Ten (SIP) and idefisk (IAX) For the > hard phone I've used Budgetone BT-102, Sipura SPA-841 and ATCOM AT-320 > (w/ IAX2 firmware). > > I've had no problems. I suppose it is a matter of a good connection. > > >>Message: 25 >>Date: Fri, 22 Jul 2005 13:48:09 +0200 (CEST) >>From: Jerry Glomph Black <[EMAIL PROTECTED]> >>Subject: Re: [Asterisk-Users] IAX over HTTP >>To: Asterisk Users Mailing List - Non-Commercial Discussion >> >>Message-ID: <[EMAIL PROTECTED]> >>Content-Type: text/plain; charset="iso-8859-1" >> >>Doing IAX over TCP is simply a Bad Idea. >> >>Under perfect circumstances, it will work OK, but the slightest >>network disturbance will result in sound gaps/distortion and/or >>monster audio delay. >> >>This is not idle UDP-boosting, I've tried it. >> >>[Have had good results with UDP-based secure tunnel transport of IAX >>traffic (CIPE and OpenVPN)] ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Extension Lights Patch
Could you pass along the information you used to get the Polycom lights to work. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Hayden Sent: Wednesday, July 20, 2005 11:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Extension Lights Patch I've been using the extension lights on my polycoms before that patch, so I'm not sure what it fixed, but I've only seen the lights work on Polycoms and Snoms. Try using the hint priority and see if it works for your gxp2000, be sure to post your results! -- Tom Hayden Astoria Telecom, LLC www.astoriatelecom.net On 7/20/05, Anton Krall <[EMAIL PROTECTED]> wrote: > Guys I just read on the wiki: > > "2005-07-19 - long awaited extension lights (hint priority) and call > pickup on various phones work with newly released asterisk patch > digium bugtracker > - feel free to test and report findings to the bugtracker to have this > commited to cvs." > > How does this work? And will it work only on certain phones or can it > work with the gxp2000? > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > -- Tom ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] voip-info.org unreliable lately?
I would also donate some bandwidth……. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian West Sent: Tuesday, June 21, 2005 9:34 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] voip-info.org unreliable lately? I would be willing to donate some bandwidth for it. We already donate bandwidth for the Asterisk CVS mirror. /b Asterisk.com/Cluecon.com On Jun 21, 2005, at 5:19 PM, Damon Estep wrote: Anyone have any insight as to why voip-info.org has been up and down all day, and more importantly unreliable for the last month? I assume the bandwidth is being donated or something, but surely someone would be willing to donate reliable bandwidth as the knowledge hosted on the site (which is also donated!) is worth way more than the bandwidth. There is no doubt it is the best documentation that exists on *, but only when accessible. Gripe, gripe, gripe… ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] NVFaxdetect
I answered my own question. I just had to dig a little deeper on the lists. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joseph Sent: Tuesday, June 21, 2005 5:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] NVFaxdetect What Linux version are you using? There is an ebuild on Gentoo -- #Joseph On Tue, 2005-06-21 at 16:15 -0500, Eric Rees wrote: > I have googled this and come up empty. Has anyone had any problems > compiling NVFaxdetect on asterisk 1.0.7? Here is the error I am > getting when I run make. > > > > > > app_nv_faxdetect.c: In function `nv_detectfax_exec': > > app_nv_faxdetect.c:210: error: structure has no member named `cid' > > app_nv_faxdetect.c:227: error: structure has no member named `cid' > > app_nv_faxdetect.c:265: error: structure has no member named `cid' > > make[1]: *** [app_nv_faxdetect.o] Error 1 > > > > > This electronic message transmission, including attachments, is for > the exclusive use of the individuals to which this e-mail is addressed > and is to be reviewed and used exclusively for authorized company > purposes. This transmission may contain proprietary, confidential or > privileged information. If you are not the intended recipient of this > transmission, you are hereby notified that any use, copying, > disclosure, dissemination, distribution or taking of any action in > reliance upon the contents of this transmission is strictly > prohibited. If you believe you may have received this electronic > message in error, please notify the sender immediately by return email > and delete or destroy the original message and/or any copy of it from > your computer system and/or your files. Thank you. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] NVFaxdetect
I have googled this and come up empty. Has anyone had any problems compiling NVFaxdetect on asterisk 1.0.7? Here is the error I am getting when I run make. app_nv_faxdetect.c: In function `nv_detectfax_exec': app_nv_faxdetect.c:210: error: structure has no member named `cid' app_nv_faxdetect.c:227: error: structure has no member named `cid' app_nv_faxdetect.c:265: error: structure has no member named `cid' make[1]: *** [app_nv_faxdetect.o] Error 1 This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1?
Correct me if I am wrong. I can remember installing a T1's with a HDSL unit at the last CO, in which the T1 was delivered to the customer's prem in two wires. I think they called this fast half-duplex. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Coulson Sent: Monday, June 13, 2005 8:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Should I choose DSL @ 1.5 or a full T1? Leon Sun wrote: > Not really true about T1 description. When you apply for T1, you need tell > vendor if it's channelized or non-ch. If you are going to use it for 1.5M > network, you need use unchannelized T1. T1 is T1. How you use the DS0s delivered across it is up to you. You can mux them out to POTS lines, use them all for data or mix it up and run voice and data over the same T1. Telco vendors don't care what you do with it, unless it's terminating for data/voice in their equipment. Even when you use all 24 channels for data, they still function as 24 distinct DS0 channels as far as timing is concerned. Unlike OC-nc circuits (Where you save some overhead for the sake of being unable to channelize the STS channels) , there is no overhead variation when channelizing a DS-1 versus using a full DS-1 for data. David -- David J. Coulson email: [EMAIL PROTECTED] web: http://www.davidcoulson.net/ phone: (216) 920-3100 / (216) 258-4942 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users This electronic message transmission, including attachments, is for the exclusive use of the individuals to which this e-mail is addressed and is to be reviewed and used exclusively for authorized company purposes. This transmission may contain proprietary, confidential or privileged information. If you are not the intended recipient of this transmission, you are hereby notified that any use, copying, disclosure, dissemination, distribution or taking of any action in reliance upon the contents of this transmission is strictly prohibited. If you believe you may have received this electronic message in error, please notify the sender immediately by return email and delete or destroy the original message and/or any copy of it from your computer system and/or your files. Thank you. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] LOOKING TO HIRE
Can we get this guy kicked off of the list. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jeffrey Richey Sent: Thursday, May 19, 2005 1:11 PM To: asterisk-biz@lists.digium.com; asterisk-users@lists.digium.com Subject: [Asterisk-Users] LOOKING TO HIRE We have positions in Ho Chi Minh City, Vietnam and Temecula, California. Please only reply to [EMAIL PROTECTED] no phone calls. Our Company comprises a diverse set of individuals who work hard and play hard. We look for motivated, dedicated candidates who have demonstrated an insatiable quest for knowledge, opportunity, responsibility and entrepreneurship. Our goals are ambitious, but ample rewards exist for those who embrace the challenge. If you are up for the challenge of helping to shape the future of an industry-leading VoIP services firm, check out our list of available positions. Vietnam Office Address: Our Company Saigon Trade Center Building 37 Ton Duc Thang Street District 1 Ho Chi Minh City, Vietnam Email: [EMAIL PROTECTED] You are welcome to e-mail your CV or resume [EMAIL PROTECTED] Available Positions in Ho Chi Minh City, Vietnam: Senior Programmer Job Description: This position requires significant technical expertise in the design and implementation of of object oriented programming and web applications. Qualified applicants will have a minimum of 8 years experience in applications development, have a thorough understanding of industry standard software development procedures and practices, and have successfully developed and implemented medium to large scale projects. Strong familiarity with the Python programming language as well as the Zope application server and Plone content management framework is required as the applications will be developed using these tools. Knowledge of HTML/CSS also beneficial. Experience working in a Unix/Linux environment is required. Basic Unix/Linux system administration skills and knowledge of MySQL database and SQL is preferred. Good spoken and written English language skills. As a Senior Programmer, work with programmers to develop the application base from specifications provided by management. Review and make technical recommendations on code developed by programmers. Mentor lower level programmer in knowledge transfer during design, build, test and implementation phase of the project. Provide System documentation for each phase of the project. Minimal Requirements: Language Requirements: Perl, PHP, MySQL; Python is a bonus Operating Systems Requirements: Linux Redhat or FREEBSD Solid knowledge of Unix based systems, TCP/IP Protocols, CVS Strong Knowledge of: www.zope.org and www.plone.org Programmer Job Description: The programmer will be responsible for implementing code in the Python language in the Zope web application server as well as standalone Python applications according to the specifications provided by management. 2+ years of Python programming experience, knowledge of SQL/MySQL, comfortable working in a Linux/Unix environment. Prior experience developing database driven web applications is a plus. Ability to speak and read/write english. Minimal Requirements: Language Requirements: Perl, PHP, MySQL; Python is a bonus Operating Systems Requirements: Linux Redhat or FREEBSD Solid knowledge of Unix based systems, TCP/IP Protocols, CVS Strong Knowledge of: www.zope.org and www.plone.org Web hosting support engineer Job Description: Responsible for providing technical support to clients, basic system administration tasks, maintaining security, and assisting the sales team with pre and post-sales support to clients in Vietnam while working with an english speaking team. Minimal Requirements: Applicants should be very familiar with the Linux operating system, Apache web server, be familiar with basic security concepts, and have some experience with at least one programming language such as PHP, Perl, or Python. Spoken and written english language skills. Web developer Job Description: The web developer will be responsible for implementing the HTML/CSS to achieve a professional look and feel for Our Company websites. The websites should adhere to W3C standards and be easily accessable to all web browsers. Note that we are not necessarily interested in flash artists or photoshop gurus. Photoshop (or even better, Gimp!) and graphic design skills will be required but HTML/CSS should really be the focus. Experience developing in Zope/Plone a plus. Ability to speak/write english. Minimal Requirements: Thorough knowledge of CSS and XHTML. Good eye for artistic design and user interface design. Project Manager Job Description: The Software Project Manager is responsible for leading a project team involved in the requirement specification, technical design, coding, integration, quality assurance test, and deployment of software projects. The Project Manager manages software projects at the managerial and project task leve
[Asterisk-Users] Broadvoice Problem
I am having problems with Broadvoice. I am not getting any audio, either in or out, but the phone will ring. Could someone double check my config. [general] context=default ; Default context for incoming calls port=5060 bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) disallow=all; First disallow all codecs allow=ulaw ; Allow codecs in order of preference register => XX:[EMAIL PROTECTED] [broadvoice] type=friend username=xx fromuser=xx authuser=xx secret= host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes insecure=yes [bv-in-1] type=user host=147.135.8.128 context=from-broadvoice dtmfmode=inband canreinvite=no nat=yes And so on for 2 and 3. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] SIP Deadlock problem.
Has anyone seen the error below or knows how to fix this. Every time this error occurs, I starting getting a 3 second delay on all internal and external calls and the only why to stop it is to stop and start asterisk. I am using a TE410 with Asterisk 1.0.7, Zaptel 1.0.7, and Libpri 1.0.7. WARNING[77191]: Avoided deadlock for 'SIP/7715-566b', 10 retries! ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Memory Requirements
MemTotal: 2074808 kB MemFree:417420 kB Buffers: 39396 kB Cached:1547124 kB SwapCached: 0 kB Active: 471180 kB Inactive: 1131508 kB HighTotal: 1179392 kB HighFree: 233536 kB LowTotal: 895416 kB LowFree:183884 kB SwapTotal: 2031608 kB SwapFree: 2031368 kB Dirty: 332 kB Writeback: 0 kB Mapped: 37696 kB Slab:43616 kB Committed_AS: 126244 kB PageTables: 1192 kB VmallocTotal: 106488 kB VmallocUsed: 3104 kB VmallocChunk: 103104 kB HugePages_Total: 0 HugePages_Free: 0 Hugepagesize: 2048 kB -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Cameron Schaus Sent: Saturday, April 09, 2005 1:37 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Memory Requirements On Fri, Apr 08, 2005 at 07:01:08PM -0500, Eric Rees wrote: > I have asterisk installed on a Dell 2850 dual-Xeon 3.0Ghz box with 2GB > of memory. This is serving about 75 sip clients, Polycom500's and > 600's. We are running into problems with the memory. Asterisk, right > now, is using about 1.8GB of system memory. I am using Asterisk 1.0.7, > Zaptel 1.0.7 with Digiums TE410 1xT1 RBS and 1xT1 PRI, Libpri 1.0.7 on > Fedora Core 3. My question; is this normal or do I need more memory or > is there a more serious underlying problem. How are measuring Asterisk memory usage? You're not counting the memory consumed by the filesystem cache, are you? Cam ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Memory Requirements
I have asterisk installed on a Dell 2850 dual-Xeon 3.0Ghz box with 2GB of memory. This is serving about 75 sip clients, Polycom500's and 600's. We are running into problems with the memory. Asterisk, right now, is using about 1.8GB of system memory. I am using Asterisk 1.0.7, Zaptel 1.0.7 with Digiums TE410 1xT1 RBS and 1xT1 PRI, Libpri 1.0.7 on Fedora Core 3. My question; is this normal or do I need more memory or is there a more serious underlying problem. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Sip registration Problems With Zyxel P2000W
You need to upgrade these phones to the latest firmware for it to work well with asterisk. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Thore Sent: Sunday, April 03, 2005 3:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Sip registration Problems With Zyxel P2000W Hi I have a Zyxel P2002 (ATA) with this config. Registration works but i cant call inn. Outgoing works fine. Any clue? Thore - Original Message - From: "Paul Dracevich" <[EMAIL PROTECTED]> To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" Sent: Sunday, April 03, 2005 6:51 AM Subject: RE: [Asterisk-Users] Sip registration Problems With Zyxel P2000W > Hi ya I have also three of these phone, here is my entry in my sip.conf > > [4701721] > type=friend > username=4701721 > secret=password721 > host=dynamic > canreinvite=no > context=internal > disallow=all > allow=g729 > dtmfmode=rfc2833 > qualify=4 > permit=0.0.0.0/0.0.0.0 > [EMAIL PROTECTED] > > > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Ugur > GUNCER > Sent: Sunday, 3 April 2005 4:37 p.m. > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Sip registration Problems With Zyxel P2000W > > Hi all, > > I bougth zyxel wifi phone but i cant register > when i want to register phone to asterisk i recieve > These errors I spend 6 hours to fix regist problem but i cant find the > solution > > [9875] > type=friend > username=9875 > secret=5789 > host=dynamic > context=default > callerid="Ugur Guncer" <9875> > canreinvite=no > dtmfmode=rfc2833 > nat=no > > > > > > > Sip read: > REGISTER sip:213.139.225.82:5060 SIP/2.0 > Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 > From: ;tag=5175B05114E474A31693 > To: > Call-ID: [EMAIL PROTECTED] > CSeq: 12 REGISTER > User-Agent: ZyXEL P2000W VoIP Wi-Fi Phone > Contact: > Expires: 300 > Content-Length: 0 > > > 10 headers, 0 lines > Using latest request as basis request > Sending to 85.99.110.143 : 43956 (non-NAT) > Transmitting (no NAT): > SIP/2.0 100 Trying > Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 > From: ;tag=5175B05114E474A31693 > To: ;tag=as369f8960 > Call-ID: [EMAIL PROTECTED] > CSeq: 12 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > Content-Length: 0 > > > to 85.99.110.143:43956 > Transmitting (no NAT): > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 85.99.110.143:43956;branch=z9hG4bK84f1504a359cd0 > From: ;tag=5175B05114E474A31693 > To: ;tag=as369f8960 > Call-ID: [EMAIL PROTECTED] > CSeq: 12 REGISTER > User-Agent: Asterisk PBX > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER > Contact: > WWW-Authenticate: Digest realm="asterisk", nonce="0f3403ce" > Content-Length: > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] ACD queue question
After I changed from leastrecent I did reload asterisk and waited about an hour and nothing changed. So I restarted asterisk and waited another hour, but it was still calling the agents in the order that they are listed in the agents.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Umar Sear Sent: Thursday, March 31, 2005 1:11 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] ACD queue question are you restarting asterisk or reloading after changing you configuration. Umar On Wed, 30 Mar 2005 19:33:42 -0600, Eric Rees <[EMAIL PROTECTED]> wrote: > I tried leastrecent. I did change the strategy, but didn't make a > difference. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Joe > Dennick > Sent: Wednesday, March 30, 2005 6:49 AM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] ACD queue question > > Using which strategy? Remember, if you change strategies and reload, > it'll forget where it was and start over. > > -Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Eric Rees > Sent: Wednesday, March 30, 2005 6:43 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] ACD queue question > > That's what I thought would happen, but after about an hour and 100 or > so incoming calls, it was still ringing the agents in the order that > they were listed in the agents.conf file. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Joe > Dennick > Sent: Tuesday, March 29, 2005 10:04 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: RE: [Asterisk-Users] ACD queue question > > The first call for each agent probably goes that way, but then after a > few calls have rolled through the queue, the strategy you specify (like > LeastRecent) should come into play. > > -Original Message- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Eric Rees > Sent: Tuesday, March 29, 2005 9:51 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: [Asterisk-Users] ACD queue question > > I have a simple 4 person ACD queue using the AgentCallback function. No > matter what strategy I use, anytime someone calls into the queue > asterisk dials the agents in the order that they are listed in the > agents.conf file. This doesn't seem right to me, or am I wrong. > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > No virus found in this incoming message. > Checked by AVG Anti-Virus. > Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 > > -- > No virus found in this outgoing message. > Checked by AVG Anti-Virus. > Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > No virus found in this incoming message. > Checked by AVG Anti-Virus. > Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 > > -- > No virus found in this outgoing message. > Checked by AVG Anti-Virus. > Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: >http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailin
RE: [Asterisk-Users] ACD queue question
That's what I thought would happen, but after about an hour and 100 or so incoming calls, it was still ringing the agents in the order that they were listed in the agents.conf file. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Joe Dennick Sent: Tuesday, March 29, 2005 10:04 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] ACD queue question The first call for each agent probably goes that way, but then after a few calls have rolled through the queue, the strategy you specify (like LeastRecent) should come into play. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Rees Sent: Tuesday, March 29, 2005 9:51 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] ACD queue question I have a simple 4 person ACD queue using the AgentCallback function. No matter what strategy I use, anytime someone calls into the queue asterisk dials the agents in the order that they are listed in the agents.conf file. This doesn't seem right to me, or am I wrong. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- No virus found in this incoming message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.8.5 - Release Date: 3/29/2005 ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] ACD queue question
I have a simple 4 person ACD queue using the AgentCallback function. No matter what strategy I use, anytime someone calls into the queue asterisk dials the agents in the order that they are listed in the agents.conf file. This doesn't seem right to me, or am I wrong. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk 1.0.6 music-on-hold
I had asterisk 1.0.5 running fine. I upgraded to 1.0.6 and now the music on hold does not work. More Detail: While I was running asterisk 1.0.5, when someone called into an Polycom IP500 and was put on hold via the Polycom "Hold" button, the hold music would play. After upgrading to 1.0.6 that does not work. But if I set up an extension to play the hold music, it plays. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Auto-Answer
That worked great. -Original Message- From: B. J. Bomar [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 01, 2005 11:20 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] Polycom Auto-Answer Take a look at http://www.voip-info.org/wiki-Cisco+7940-7960+auto-answer+config. With a few small modifications it should work like a champ on the Polycom phones. B. J. -Original Message----- From: Eric Rees [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 01, 2005 10:38 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [Asterisk-Users] Polycom Auto-Answer That's kind of what I thought, but I am trying to put together a phone to multi-phone paging system. I all ready have and overhead paging systems, but the powers-at-be want a phone paging system. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 01, 2005 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom Auto-Answer Eric Rees wrote: > I am having a problem with Polycom auto-answer. I have the auto-answer > working between PhoneA and PhoneB, but when I try to use the intercom > between more then one phone I start having problems. PhoneA dials *3 > which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only > one will pick up, the rest will hang up and I get this error on > Asterisk: Got SIP response 500 "Internal Server Error". U Yeah. What did you think was going to happen, Asterisk was going to magically bridge four phones together because they all answered? As soon as one phone answers, the call is complete and the remaining phones will not be able to answer (because the calls going out to them will have been destroyed). If you need more than two parties in a call, you need to use MeetMe or one of the other conferencing applications. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Polycom Auto-Answer
That's kind of what I thought, but I am trying to put together a phone to multi-phone paging system. I all ready have and overhead paging systems, but the powers-at-be want a phone paging system. -Original Message- From: Kevin P. Fleming [mailto:[EMAIL PROTECTED] Sent: Tuesday, March 01, 2005 10:13 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Polycom Auto-Answer Eric Rees wrote: > I am having a problem with Polycom auto-answer. I have the auto-answer > working between PhoneA and PhoneB, but when I try to use the intercom > between more then one phone I start having problems. PhoneA dials *3 > which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only > one will pick up, the rest will hang up and I get this error on > Asterisk: Got SIP response 500 "Internal Server Error". U Yeah. What did you think was going to happen, Asterisk was going to magically bridge four phones together because they all answered? As soon as one phone answers, the call is complete and the remaining phones will not be able to answer (because the calls going out to them will have been destroyed). If you need more than two parties in a call, you need to use MeetMe or one of the other conferencing applications. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom Auto-Answer
I am having a problem with Polycom auto-answer. I have the auto-answer working between PhoneA and PhoneB, but when I try to use the intercom between more then one phone I start having problems. PhoneA dials *3 which calls PhoneB, PhoneC, and PhoneD. All the phones ring, but only one will pick up, the rest will hang up and I get this error on Asterisk: Got SIP response 500 "Internal Server Error". ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk and Fedora Core 3
http://www.voip-info.org/tiki-index.php?page=Asterisk+Fedora+Core+3 Use the wiki luke. -Original Message- From: Bill Maidment [mailto:[EMAIL PROTECTED] Sent: Thursday, February 10, 2005 5:08 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Asterisk and Fedora Core 3 Hi guys I'm new to this list and I imagine this question has been asked before, so feel free to point me to the correct references. My question is, how do you install asterisk on Fedora Core 3, with all rpm updates, seeing as there is no kernel-source rpm anymore? Thanks for any advice. -- _/_/_/_/ _/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/_/ _/ _/ _/ _/_/_/_/ _/ _/ _/ Bill Maidment Maidment Enterprises Pty Ltd Unless you are named "Alfred E. Newman", you may read only the "odd numbered words" (every other word beginning with the first) of the message above. If you have violated that, then you hereby owe the sender AU$10 for each even numbered word you have read. Adapted from "Stupid Email Disclaimers" (see http://www.goldmark.org/jeff/stupid-disclaimers/) ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] TDM400 Problem
Has anyone seen this message trying to install an TDM400.. spurious 8259A interrupt: IRQ7 This error happens after I do a modprobe wctdm and then the system hangs. I am installing this in an Asus motherboard with a VIA P4M266 chipset. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Re: Polycom and call waiting again..
Thanks for the heads up. I guess I will have to start looking into
Setgroup and Checkgroup.
-Original Message-
From: Jon Radon [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 27, 2005 12:07 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Polycom and call waiting again..
incominglimit is deprecated. It will be EOL'd.
http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+incominglimit
On Thu, 27 Jan 2005 10:21:25 -0600, Eric Rees <[EMAIL PROTECTED]>
wrote:
> Here is what I have done to get around the call waiting problem.
> This is for a Polycom 500. This is kind of a pain, but it works.
>
> Exten.conf
> exten => 1051,1,Dial(SIP/1051,20,tTr)
> exten => 1051,2,Voicemail(u${EXTEN})
> exten => 1051,102,Dial(SIP/1051b,20,tTr)
> exten => 1051,103,Dial(SIP/1051c,20,tTr)
> exten => 1051,104,Voicemail(b${EXTEN})
>
> Sip.conf
> [1051]
> type=friend
> username=1051c
> callerid="NMS001"<1051>
> host=dynamic
> dtmfmode=rfc2833
> mailbox=1051
> context=sip
> callgroup=1
> pickupgroup=1
> canreinvite=no
> imcominglimit=1
> [1051b]
> type=friend
> username=1051c
> callerid="NMS001"<1051>
> host=dynamic
> dtmfmode=rfc2833
> mailbox=1051
> context=sip
> callgroup=1
> pickupgroup=1
> canreinvite=no
> imcominglimit=1
> [1051c]
> type=friend
> username=1051c
> callerid="NMS001"<1051>
> host=dynamic
> dtmfmode=rfc2833
> mailbox=1051
> context=sip
> callgroup=1
> pickupgroup=1
> canreinvite=no
> imcominglimit=1
>
> -Original Message-
> From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
> Sent: Thursday, January 27, 2005 9:28 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Re: Polycom and call waiting again..
>
> Adam Goryachev wrote:
>
> > [local-stuff]
> > ; This is where we pretend a channel is an extension
> >
> > exten => 1234,1,SetGroup(SIP1234)
> > exten => 1234,2,CheckGroup(1)
> > exten => 1234,3,Dial(SIP/1234,15)
> > exten => 1234,104,Busy
> >
> > [queue-stuff]
> > exten => 6939,1,AddQueueMember(Local/${CALLERIDNUM})
>
> You are close... that should be:
>
> AddQueueMember(Local/[EMAIL PROTECTED])
>
> That way when the queue app tries to call the agent, it will have an
> extension _and_ a context to deliver the call to.
> ___
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> ___
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> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
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> http://lists.digium.com/mailman/listinfo/asterisk-users
>
--
Is it something someone said, was it something someone said?
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RE: [Asterisk-Users] Re: Polycom and call waiting again..
Here is what I have done to get around the call waiting problem.
This is for a Polycom 500. This is kind of a pain, but it works.
Exten.conf
exten => 1051,1,Dial(SIP/1051,20,tTr)
exten => 1051,2,Voicemail(u${EXTEN})
exten => 1051,102,Dial(SIP/1051b,20,tTr)
exten => 1051,103,Dial(SIP/1051c,20,tTr)
exten => 1051,104,Voicemail(b${EXTEN})
Sip.conf
[1051]
type=friend
username=1051c
callerid="NMS001"<1051>
host=dynamic
dtmfmode=rfc2833
mailbox=1051
context=sip
callgroup=1
pickupgroup=1
canreinvite=no
imcominglimit=1
[1051b]
type=friend
username=1051c
callerid="NMS001"<1051>
host=dynamic
dtmfmode=rfc2833
mailbox=1051
context=sip
callgroup=1
pickupgroup=1
canreinvite=no
imcominglimit=1
[1051c]
type=friend
username=1051c
callerid="NMS001"<1051>
host=dynamic
dtmfmode=rfc2833
mailbox=1051
context=sip
callgroup=1
pickupgroup=1
canreinvite=no
imcominglimit=1
-Original Message-
From: Kevin P. Fleming [mailto:[EMAIL PROTECTED]
Sent: Thursday, January 27, 2005 9:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Re: Polycom and call waiting again..
Adam Goryachev wrote:
> [local-stuff]
> ; This is where we pretend a channel is an extension
>
> exten => 1234,1,SetGroup(SIP1234)
> exten => 1234,2,CheckGroup(1)
> exten => 1234,3,Dial(SIP/1234,15)
> exten => 1234,104,Busy
>
> [queue-stuff]
> exten => 6939,1,AddQueueMember(Local/${CALLERIDNUM})
You are close... that should be:
AddQueueMember(Local/[EMAIL PROTECTED])
That way when the queue app tries to call the agent, it will have an
extension _and_ a context to deliver the call to.
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[Asterisk-Users] Polycom Call-Waiting
Has anyone been able to find a way to disable call-waiting on Polycom phones? ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Zap Channel Group Question
I have a channelized T1 with the first 12 channels set to FXS_GS. In my extension.conf file, I have a variable in [globals] DIALOUT=ZAP/g1. The problem is when I try to make an outbound call, the console tells me that everything is busy, but is I change the variable to DAILOUT=ZAP/1, I can dial out no problem. Here is my Zapata.conf: ;outbound context=default signalling=fxs_gs group=1 channel=>1-12 The DIALOUT/g1 worked with this was connected to a PRI. Is the group command different when set to FXS_GS? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] E&M Wink Question
List: I already have asterisks up and running on a PRI, but where we are moving we cannot get a PRI so we are going to get T1. My question is: We are going to us E&M Wink for signaling with DTMF and caller id. The channels are going to be setup like this, 12 channels for 2-way and 12 channels for incoming only with DIDs. How would I configure the zaptel.conf? I realize that I will have two groups, one for incoming and another for incoming and outgoing. I have setup asterisks for PRI's but this will be the first non-PRI install for me. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Is this possible
That does part of what I want, but the callerid isn't showing what I need. It shows incoming call from "WhoEver" instead of call from "WhoEver" to "WhatEver" on the assistants phone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Sean Kennedy Sent: Monday, December 06, 2004 1:44 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Is this possible Eric Rees wrote: >I don't know if this is possible, so I will let the collective decide. > >Here is what I would like to do. > >BossA calls BossB, BossB's admin assistant sees the call from BossA on >her phone. CallerID would look something like: BossA to BossB : on her >phone. And she would be able to pick if BossB was not in his office. I >am sure this is possible, but I do not know where to start, or even how >to search on this. > > Have the phone call ring both extensions. Depending on the phone, you could even set up the assistant's phone to have a special ring for this situation ( or, indeed, no ring ). ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Is this possible
I don't know if this is possible, so I will let the collective decide. Here is what I would like to do. BossA calls BossB, BossB's admin assistant sees the call from BossA on her phone. CallerID would look something like: BossA to BossB : on her phone. And she would be able to pick if BossB was not in his office. I am sure this is possible, but I do not know where to start, or even how to search on this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Umlaut over I on Definity display
I have a similar setup, and when get the same thing displayed on our 6408D+ phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Miedema, Bud Sent: Friday, December 03, 2004 1:40 PM To: '[EMAIL PROTECTED]' Subject: [Asterisk-Users] Umlaut over I on Definity display I've successfully integrated * to a Definity G3SI PBX via PRI. On calls from the * box to a Definity display telephone a umlaut over an I appears at the beginning of the caller name on the Definity display. Anyone seen this before? Thanks... Bud ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Problem or Polycom Problem
Neither one is turned on. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling aka ManxPower Sent: Thursday, December 02, 2004 6:32 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Asterisk Problem or Polycom Problem Eric Rees wrote: > Thanks for you suggestion, but the last time I tried this I was talking > to a person and it cut me off. But I will try what you suggested. If you have busydetecr or callprogress in zapata.conf, turn them off. --Eric -- I am seeking part or full time employment in the Greater Toronto Area, My preference is part time employment with some telecommuting, but all offers will be considered. Contact eric at fnords.org. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Problem or Polycom Problem
Thanks for you suggestion, but the last time I tried this I was talking
to a person and it cut me off. But I will try what you suggested.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Goryachev
Sent: Thursday, December 02, 2004 5:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Asterisk Problem or Polycom Problem
On Fri, 2004-12-03 at 05:35, Eric Rees wrote:
> We are in the process of testing * for company wide deployment. We
> are using Polycom 300 phones, the only problem that I am running into
> is when I call an 800 number that has an IVR I get disconnected after
> about 60 seconds. Here are the logs from asterisk. I am not sure if
> this is a problem with asterisk timing out or if it is the phone. To
> me this looks like asterisk is timing out.
>
>
>
> Executing Dial("SIP/1001-058c", "Zap/g1/91877xxx") in new stack
>
> -- Called g1/91877xxx
>
> -- Hungup 'Zap/1-1'
>
> == Spawn extension (sip, 91877xxx, 1) exited non-zero on
> 'SIP/1001-058c'
I was reading through the polycom admin manual to fine-tune a customers
polycom 300 and 600 phones, when I saw some settings. the default ring
time (ie, before answer) is 60 seconds. Consider that your call is
probably not answered until the real person answers, so the polycom is
probably giving up and hanging up the call. This can be modified from
the ipmid.cfg file and I think can be set to 0 to wait for 'infinity'.
I can't remember which xml tag it is, but I think it would be in the
call section.
Regards,
Adam
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RE: [Asterisk-Users] Ring all Configured Extension
Where only talking about 100 extensions. That is a lot to hard code by hand. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Thursday, December 02, 2004 12:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Ring all Configured Extension Why are you afraid of that suggestion? Matthew - Original Message - From: "Eric Rees" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Thursday, December 02, 2004 10:56 AM Subject: RE: [Asterisk-Users] Ring all Configured Extension I was afraid that someone would suggest that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Thursday, December 02, 2004 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Ring all Configured Extension exten => 4000,1,Dial(SIP/3001&SIP/3002&SIP/3003&..., 30, t) Matthew - Original Message - From: "Eric Rees" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Thursday, December 02, 2004 8:56 AM Subject: [Asterisk-Users] Ring all Configured Extension I don't know if the is possible on not. I would like to know the easiest way to ring all extensions in the sip.conf file for intercoms. I have phone to phone intercom working. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk Problem or Polycom Problem
We are in the process of testing * for company wide
deployment. We are using Polycom 300 phones, the only problem that I am
running into is when I call an 800 number that has an IVR I get disconnected
after about 60 seconds. Here are the logs from asterisk. I am not
sure if this is a problem with asterisk timing out or if it is the phone.
To me this looks like asterisk is timing out.
Executing Dial("SIP/1001-058c", "Zap/g1/91877xxx")
in new stack
-- Called g1/91877xxx
-- Hungup 'Zap/1-1'
== Spawn extension (sip, 91877xxx, 1) exited
non-zero on 'SIP/1001-058c'
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RE: [Asterisk-Users] Ring all Configured Extension
I was afraid that someone would suggest that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Matthew Boehm Sent: Thursday, December 02, 2004 10:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Ring all Configured Extension exten => 4000,1,Dial(SIP/3001&SIP/3002&SIP/3003&..., 30, t) Matthew - Original Message ----- From: "Eric Rees" <[EMAIL PROTECTED]> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <[EMAIL PROTECTED]> Sent: Thursday, December 02, 2004 8:56 AM Subject: [Asterisk-Users] Ring all Configured Extension I don't know if the is possible on not. I would like to know the easiest way to ring all extensions in the sip.conf file for intercoms. I have phone to phone intercom working. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Ring all Configured Extension
I don't know if the is possible on not. I would like to know the easiest way to ring all extensions in the sip.conf file for intercoms. I have phone to phone intercom working. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Spandsp kind of working
I have spandsp installed and working, but when it emails using Scotts mailfax, the attachment is a dat file. I tried to rename the file to .tiff or .pdf, but it will not open. In the /var/spool/asterisk/fax folder, that faxes are there as tiffs, and I can open those without any trouble. The problem is in the conversion from tiff to pdf. Is there another package that needs to be installed for the conversion to work? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Spandsp and Asterisk
I was finaly able to patch the Makefile in the apps dir. I used 2pre4 version. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Seth Remington Sent: Tuesday, November 23, 2004 8:22 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Spandsp and Asterisk On Tue, 2004-11-23 at 09:00, Eric Hall wrote: > Does anyone have an update patch file to get Spandsp installed? > > I'm running asterisk CVS-HEAD-11/19/04-21:53:37 on redhat 9.0 > I installed spandsp-0.0.2 > > > when runnig the patch I get > > patching file Makefile > Hunk #1 FAILED at 41. > Hunk #2 FAILED at 69. > 2 out of 2 hunks FAILED -- saving rejects to file Makefile.rej > Make sure you are trying to patch the Makefile in the apps directory, not the top-level Makefile. -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Patching asterisk for spandsp
I realized that after this first two times I tried that, but I still will not patch. I tried to path the file manually. This is where make clean dies at. app_rxfax.so : app_rxfax.o $(CC) $(SOLINK) -o $@ $< -lspandsp -ltiff app_txfax.so : app_txfax.o $(CC) $(SOLINK) -o $@ $< -lspandsp -ltiff The first part of the patch works, but the second does not. I am using the latest CVS of asterisk and spandsp. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tom Neville Sent: Monday, November 22, 2004 3:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Patching asterisk for spandsp I think I found the problem with the patch.. are you applying that patch to the Makefile in your asterisk source directory?? or to the Makefile in the asterisk/apps directory? I got the same error until I applied it against the asterisk/apps/Makefile. Tom On Nov 22, 2004, at 4:15 PM, Steve Prior wrote: > Gregory Junker wrote: > >> Just for sanity's sake, I went back and read the README on the site >> again, and it does say: >> "Add the files rxfax.c, txfax.c and dtmftotext.c (the last one has >> nothing to do with the fax machine, but my makefile patch expects it >> to be present)" >> You have to grab the dtmftotext.c file as well, which also is not >> part of the tarball. That could be the problem. >> Greg > > I found the dtmftotext.c wasn't needed for the most recent version of > spandsp on the ftp site, and I also didn't have to modify the Makfile > with the path where to put the fax files - it is now set in > extensions.conf > instead. Examples of this are at scott's site in my original note on > this. > > Steve > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Patching asterisk for spandsp
I tried using the link you provided. This is the error it gives me. patching file ../Makefile Hunk #1 FAILED at 48. Hunk #2 FAILED at 77. 2 out of 2 hunks FAILED -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Seth Remington Sent: Monday, November 22, 2004 2:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Patching asterisk for spandsp On Mon, 2004-11-22 at 14:38, Eric Rees wrote: > When I try to patch the Makefile for asterisk with the > Apps_makefile.patch from Spandsp I get the following error. > > patching file Makefile > Hunk #1 FAILED at 47. > Hunk #2 FAILED at 76. > 2 out of 2 hunks FAILED I haven't updated this in a while but you can try it and see if it works... http://sremington.zapto.org/downloads/asterisk/spandsp/Makefile.patch -Seth -- Seth Remington SaberLogic, LLC 661-B Weber Drive Wadsworth, Ohio 44281 Phone: (330)335-6442 Fax: (330)336-8559 ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Patching asterisk for spandsp
When I try to patch the Makefile for asterisk with the Apps_makefile.patch from Spandsp I get the following error. patching file Makefile Hunk #1 FAILED at 47. Hunk #2 FAILED at 76. 2 out of 2 hunks FAILED Has anybody seen this. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] [Asterisk-User] recommendation for IP phones
Polycom phones are nice and are about half the cost of Cisco phone. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Kavit Munshi Sent: Thursday, November 18, 2004 9:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] [Asterisk-User] recommendation for IP phones Can any one recommend IP phones that work the best with asterisk and dont cause a major dent in your finances? I shall be using them for normal office functions nothing out of the ordinary. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Polycom 300 registration
We are having a problem with the Polycom 300. For some reason, it will deregister and not register back. I have looked the config files for the Polycom, but since it is all XML I might be missing something. Thanks. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
