RE: [asterisk-users] SIP Options Reply Ignored
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: 03 June 2007 18:30 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] SIP Options Reply Ignored On Sun, 3 Jun 2007, Ian Clough wrote: The problem comes when I try to use a SIP phone at home (also behind a NAT router). The phone registers correctly and I can see the SIP OPTONS packets being sent to the phone (SNOM 190). I can see an OK reply being received by Asterisk (using SIP DEBUG). However the OK reply appears to be ignored and a retransmission is made and the phone is marked as UNREACHABLE and will not accept any calls. Alex Wait, so this is the phone registering to Asterisk? Any inconsistencies Alex in the source/destination ports vis-a-vis the NAT state pinholes? -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 Yes once the phone registers OK asterisk sends the OPTIONS packets (qualify=yes) This is an example. It shows asterisk reading a reply from by phone to transmission #3 and then sending retransmission #4 intechdial*CLI --- SIP read from xxx.xxx.xxx.xxx:2057 --- SIP/2.0 200 OK Via: SIP/2.0/UDP 207.13.251.93:5060;branch=z9hG4bK24ffe501;rport=40804 From: asterisk sip:[EMAIL PROTECTED];tag=as35e434f3 To: sip:[EMAIL PROTECTED]:2057;line=15aykp4d Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS Contact: sip:[EMAIL PROTECTED]:2057;line=15aykp4d User-Agent: snom190/3.60x Accept-Language: en Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Content-Length: 0 - --- (14 headers 0 lines) --- Retransmitting #4 (NAT) to xxx.xxx.xxx.xxx:2057: OPTIONS sip:[EMAIL PROTECTED]:2057;line=15aykp4d SIP/2.0 Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bK24ffe501;rport Max-Forwards: 70 From: asterisk sip:[EMAIL PROTECTED];tag=as35e434f3 To: sip:[EMAIL PROTECTED]:2057;line=15aykp4d Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX SVN-trunk-r63567 Date: Thu, 31 May 2007 11:01:26 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Supported: replaces Content-Length: 0 Where xxx.xxx.xxx.xxx is the external address of my home router Yyy.yyy.yyy.yyy is the external address of the office router 207.13.251.93 is the internal address of the asterisk server 192.168.1.50 is the internal address of the phone Ian C ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] click to call
On Sun, Jun 03, 2007 at 11:29:57PM -0500, Anton Krall wrote: True, maybe I didnt make myself clear on that point, what i meant was, Im not looking for an app that would let people click a sip: URL type to make a call using their already installed softphone but rather allow any user that visits our website What assumptions are of any user? * Java? * Flash? * ActiveX? * Pure javascript? * Any other technology? A specific dialect of one of the above? Obviously with pure HTML this is impossible. Almost obviously that with pure javascript as well. to click on something and either open a web softphone or download a small one thats preconfigured and allow them to call a predefined extension on our asterisk server. And use a preconfigured sound card on their local system. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel on CENTOS servercd
One comment if I may: http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos does need some cleanup. Let alone merging of the two separate procedures and an update. Also: On Sun, Jun 03, 2007 at 06:22:57PM -0400, Lee Jenkins wrote: Khaled Chehab wrote: I suffered a lot from installing zaptel 1.4.2 on centos servercd ,Sure there is someone did that, Please in need to have the installation procedure step by step. Its too urgent for me . Thanks alot These scripts are working pretty well for me. They are based on a script I found on the wiki. # Script to download pre reqs for installation # == #!/bin/bash rpm --import http://mirror.centos.org/centos/RPM-GPG-KEY-CentOS-4 Isn't this done by the installer? yum -y install kernel-source bison openssl-devel Why do you need kernel-source? I know it is fun do download some extra 40Gigs, but apart from that, the kernel-devel / kernel-smp-devel is all you need to build zaptel. (likewise linux-headers-`uname -r` on Debian/Ubuntu ) yum -y update # Use kernel-devel instead of kernel-smp-devel # if not a SMP machine # |==| | | yum -y install gcc kernel kernel-smp-devel bison openssl-devel Hmmm you install kernel, but kernel-*smp*-devel . Strange ocmbination. Why not simply 'yum update' to get the latest kernel, if needed? This will upgrade either 'kernel' or 'kernel-smp', depending on the installed package. If you don't need a kernel upgrade you don't need the following: shutdown -r now # end script # Script to download asterisk, etc and build/install # == #!/bin/bash rm -f /usr/lib/asterisk/modules/* cd /usr/src rm -rf asterisk wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.18.tar.gz tar -zxvf asterisk-1.2.18.tar.gz mv asterisk-1.2.18 asterisk cd /usr/src rm -rf zaptel wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.17.1.tar.gz tar -zxvf zaptel-1.2.17.1.tar.gz mv zaptel-1.2.17.1 zaptel cd /usr/src rm -rf asterisk-addons wget http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.2.6.tar.gz tar -zxvf asterisk-addons-1.2.6.tar.gz mv asterisk-addons-1.2.6 asterisk-addons cd /usr/src rm -rf asterisk-sounds wget http://ftp.digium.com/pub/asterisk/releases/asterisk-sounds-1.2.1.tar.gz tar -zxvf asterisk-sounds-1.2.1.tar.gz mv asterisk-sounds-1.2.1 asterisk-sounds cd /usr/src/zaptel make clean make install make config cd /usr/src/asterisk make clean make install make samples make config cd /usr/src/asterisk-addons make clean make install cd /usr/src/asterisk-sounds make install -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] nvlinedetect for Asterisk 1.4
Hi All, I need nvlinedetect and installation guide for my asterisk server version 1.4, I appreciate if some one can help me on this. I have spent so much time to search google for the answer but no luck. The only solution that I found is nvlinedetect (0.9) for asterisk 1.2. I encountered compilation error with nvlinedetect 0.9 on my asterisk 1.4 ASLAY ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] click to call
On 3 Jun 2007, at 03:37, Steve Totaro wrote: -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Dean Collins Sent: Saturday, June 02, 2007 10:04 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] click to call -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, 2 June 2007 9:41 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: RE: [asterisk-users] click to call Then there is risk that the call quality will be poor which will make your company look bad as well. Anyways, I prefer email over the phone. Funny huh? Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB Hmm yes Skype quality very bad, hmmm packet 8, vonage and every other voip provider very bad. Dude give it a rest. Regards, Dean Collins Cognation Pty Ltd [EMAIL PROTECTED] +1-212-203-4357 Ph +61-2-9016-5642 (Sydney in-dial). Again, you are selling a product and have an agenda Dude. Sales 101, create demand in the customer's mind (even when there really is not any in reality). The only apples to apples comparison you made is Skype and they do not use Java or IAX that I am aware of. Packet8 and Vonage are completely irrelevant and just your way of spinning the issue in your direction. Welcome to the no spin zone. Like I said, OHHH cool! I can click this button and call. Let's see some usage stats. The proof is the pudding. ROI? Usage? Customer impressions? Until you can provide these details, then I think it is you who should give it a rest. Thanks, Steve As a point of fact, Dean is no longer speaking as a rep of Mexuar (check his .sig) Mexuar's official position is to recognize that in many use-cases there will be a significant number of uses who won't be able or willing to use a softphone (browser based or not). Solution providers will need to provide 'conventional' call back solutions alongside the pure voip style. In our 'Corraleta connect' hosted solution we do exactly that. There are some demographics where this isn't necessary, it is up to the solution provider to identify this requirement - not us as a technology provider. Tim Panton www.mexuar.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] click to call
On 4 Jun 2007, at 08:14, Tzafrir Cohen wrote: On Sun, Jun 03, 2007 at 11:29:57PM -0500, Anton Krall wrote: True, maybe I didnt make myself clear on that point, what i meant was, Im not looking for an app that would let people click a sip: URL type to make a call using their already installed softphone but rather allow any user that visits our website What assumptions are of any user? * Java? We are seeing _very_ high numbers for users with Java 1.4 installed * Flash? * ActiveX? * Pure javascript? Again most desktop browsers support javascript. * Any other technology? A specific dialect of one of the above? Obviously with pure HTML this is impossible. Almost obviously that with pure javascript as well. Yep, I have never seen an audio capture interface for javascript. to click on something and either open a web softphone or download a small one that’s preconfigured and allow them to call a predefined extension on our asterisk server. And use a preconfigured sound card on their local system. For 'consumer' OS's like windows and OSX that's pretty much the norm. For Linux it isn't (yet), it varies from distribution to distribution. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mixing Vars into Voicemail WAVs
Has anyone out there tried to mix the envelope metadata for voicemails into the audio payload that's stored by Asterisk? I would like to have the CID and Timestamp baked into the beginning of the WAV file, not just as text in the email itself. Thanks! -Rob. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
On 4 Jun 2007, at 01:00, Andrew Kohlsmith wrote: On Sunday 03 June 2007 4:30 pm, Alex Crow wrote: No frills, specs look good, price seems excellent! http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519 That's a terrible phone. I've tried them. the screen is pretty much useless, the buttons are *TINY*, the battery life horrible, and the ringtones gimmicky. I haven't tried WEP or WPA on these things, but the phones I've gotten rid of long ago due to their problems. -A. If you can face the configuration hell, the Nokia e60 should do what you want Tim. Tim Panton www.mexuar.net www.westhawk.co.uk/ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING
Hi all, sorry if i have missed something, i was just curious what unicall actually is, what the main features are and did a quick search on voip-info.org and plain old google but didn't find some source of information that simply says : Unicall is this and does that ... Can anyone point me to some source of information ? Thanks a lot ... Tobias Wolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Digium Card
HI I'm looking for a card that support both PRI and TDM. Please suggest me ? thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Issue with Grandstream ATA 496.
Hi everybody, I have bought one 496 to use as ATA for two analog extensions in my office. I'm experiencing strange behaviuors. The ATA blocks itself and needs to be rebooted. Sometimes it hungs the lines(ISDN bristuffed HFC single isdn line). It was update to last available firmware. Has anybody those issues? Best regards Mauro ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Wireless IP Phone with external Telephone Book
Hi, we are searching for wireless IP Phones (DECT preferred) with have an solution for an external telephone book. We don't want to enter all of our numbers into every telephone, but have one location for all the numbers and every phone looks them up there, e.g. an ldap server. We have tried Kirk but they are working on an solution without any information when it will be available. Does anyone knows an vendor which supports this feature ? Regards, Tobias Wolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Card
On Mon, Jun 04, 2007 at 01:08:15PM +0400, Arun Kumar wrote: HI I'm looking for a card that support both PRI and TDM. Please suggest me ? Non such card. You need two cards. Why exactly do you need a single card? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Cutted audio or 2/3s blankson EuroISDN- Asterisk1.4
We're running 1.4.0 of asterisk 1.4.2.1 of zaptel And kernel 2.6.20-1.2316.fc5smp The information contained in this email is intended for the personal and confidential use of the addressee only. It may also be privileged information. If you are not the intended recipient then you are hereby notified that you have received this document in error and that any review, distribution or copying of this document is strictly prohibited. If you have received this communication in error, please notify Brendata immediately on: +44 (0)1268 466100, or email '[EMAIL PROTECTED]' Brendata (UK) Ltd Nevendon Hall, Nevendon Road, Basildon, Essex. SS13 1BX UK Registered Office as above. Registered in England No. 2764339 See our current vacancies at www.brendata.co.uk ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING
HI Tobias, look in www.soft-switch.org/unicall/unicall/index.html ;p -- Humberto Figuera - Using Linux 2.6.20 Usuario GNU/Linux 369709 Caracas - Venezuela GPG Key Fingerprint = 5AAC DF0C 00F4 2834 28BA 37AD 3364 01D1 74CA 0603 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] yum om centos
I have 2 servers, one connected to internet and the other is on a private lan have no access to internet. On the first server I update the kernel by yum update And installed asterisk prerequisite module yum install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf \ libtool make automake automake14 automake15 automake16 automake17 \ bison byacc flex libtermcap libtermcap-devel newt newt-devel \ ncurses ncurses-devel openssl-devel zlib zlib-devel krb5-devel I zipped /var/cache/yum from the first server and extract it on the second server at the same directory. On the second server I tried to update using yum update but the yum update failed. How can I do that with out connecting the second server to internet . Khaled Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] click to call
On Sun, 3 Jun 2007, Anton Krall wrote: Hi Gordon So, mexuar solution was that java softphone that you talked about? Yes. Any other small softphone type solution around, something on the same lines of what you described, something that the user could download but could be preconfigured or passed parameters to so they user wont have to mess with settings. When looking for something else, I found that you can get a version of x-lite which can be dynamically re-configured at download time - Sipgate (www.sipgate.co.uk) use this. Once logged into their site, you download their version of x-lite (for free) and it just works on your PC, as they've included the account, IP address, etc. details as part of the download. Cost (from x-lite) was about the same ~£1000 for the setup fee and tools required (from memory). The down-side of that version of x-lite is that it's windows only (Although the generic x-lite is multi platform) I looked at idefisk (my soft-phone of choice), but I wasn't sure if that too could be dynamically configured at download time. (Again about the same cost to have your own custom skin on it, etc.) These both end up as installed programs though - not transient like a java based one. Reading through these threads, it's clear to me that there is confusion as to what click to call really means (for some people). I think the issue is simply that there is more than one way to do it! I think the OP in this thread wanted something along the lines of what I described, but the other (or classic way?) of click to call would involve the web visitor entering in their phone number on the web page, pushing click to call, then it initiating a call via (eg) a .call file being placed in the asterisk call spool directory, so their phone rings, they pick it up, then the agents phone rings... and they talk to each other, with the agent footing the bill as it's then a PSTN call. The Java method is a pure VoIP method, so might be more attractive in some cases... But I'd be glad to enter one of my 0909[1] numbers into a click-to-call box ;-) Gordon [1] Calls cost the sender £1.50 a minute, and the owner of the number can get a substantial chunk of that!___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
On Sun, 3 Jun 2007, Andrew Kohlsmith wrote: On Sunday 03 June 2007 4:30 pm, Alex Crow wrote: No frills, specs look good, price seems excellent! http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519 That's a terrible phone. I've tried them. the screen is pretty much useless, the buttons are *TINY*, the battery life horrible, and the ringtones gimmicky. I have to disagree on at least one point here - Battery life. I don't think 3 or 4 days standby and several hours of talk time makes for a horrible battery life. The F1000G has other faillings, but battery life isn't one of them! I haven't tried WEP or WPA on these things, but the phones I've gotten rid of long ago due to their problems. Mine works with both WEP and WPA. It just occasionally won't re-attach to an access point (iwthout rebooting) and won't roam between access points at all. I think for the cost (and you can get them cheaper than from Scan!) they're not bad, but still in the experimental/toy category than something I'd deploy to clients. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] yum om centos
independently install each rpm via rpm command :-/ On 04/06/07, Khaled Chehab [EMAIL PROTECTED] wrote: I have 2 servers, one connected to internet and the other is on a private lan have no access to internet. On the first server I update the kernel by yum update And installed asterisk prerequisite module yum install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf \ libtool make automake automake14 automake15 automake16 automake17 \ bison byacc flex libtermcap libtermcap-devel newt newt-devel \ ncurses ncurses-devel openssl-devel zlib zlib-devel krb5-devel I zipped /var/cache/yum from the first server and extract it on the second server at the same directory. On the second server I tried to update using yum update but the yum update failed. How can I do that with out connecting the second server to internet . Khaled Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Digium Card
Tzafir Most small/medium companies have a T1 for all their phone needs. Internally there is a need for some analog lines. * Fax Machine - FXS * Security System (most ask/demand two lines) FXS * Paging - FXO * Dialup systems Andrew On 6/4/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Jun 04, 2007 at 01:08:15PM +0400, Arun Kumar wrote: HI I'm looking for a card that support both PRI and TDM. Please suggest me ? Non such card. You need two cards. Why exactly do you need a single card? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: click to call
I am using the free http://www.babarnazmi.citril.com/forum/viewtopic.php?t=7sid=fd8047cffb13074969d3418064f4eb31 It is working as you described. It appears to be working well. -- -- Steven http://www.glimasoutheast.org Anton Krall [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] The idea is to put some kind of embedded app on the website so customers with mics can just click an icon or image and connect to our sales people or customer support staff... So far for what I've seen, there is some misconception of the terms.. click to dial can mean if you see a number on a webpage, click on it and your softphone will dial it.. but can also mean click on the image and it will connect you to the sales people, for example. I'm looking for the latter. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Jueves, 31 de Mayo de 2007 10:18 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] click to call Anton Krall wrote: I have been looking around for examples or code on making a click to call application for web sites... has anybody had any luck on this topic? Is there any open source code out ther that could do this? What we have done in the past is created url's like this : sip:4044565941. Xlite will register itself as the sip handler on your system. If you want a generic click to call (ability to call numbers on any given website) check out moziax ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
On Monday 04 June 2007 8:24 am, Bryan Laird wrote: - Physically the phone feels very light and cheap, that if you were to drop it that it might not survive very many of them. The buttons feel more like a toy than anything else but once you get beyond that it works. How are they for big hands? I'll have to do some checking around to see if I can find a rubber case for it or something, it's all concrete floors here. - Address book storage is ok the interface from the phone is fairly standard for what you would see in a cell phone and adding entries isn't really all that horrid of a task. You can also add entries via the web interface which does make for an easier way to add several entries but the lack of anything resembling a 'sync' function could be considered bothersome. Bugger. Last thing, one neat thing about the wip300 if you are adventurous is the fact that the firmware is under GPL... so if you really felt like it you could probably change the behavior of the phone. This I was not aware of. I will certainly evaluate this phone and it's bigger brother. Anyway sorry for the long message but I felt like chiming in on this. All in all I don't think it's a horrible phone I do however think it's over priced for what it is but not enough demand on this type of device is always going to keep the price up in the air. Your message is *exactly* the kind of reply I was hoping to get. Thank you so much for taking the time to write such a long response. I truly appreciate it. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless IP Phone with external Telephone Book
Not perfect solution but aastra sets support xml. Which then can do server based directory. If you can wait 2 weeks or so, I should be able to tell you how welworks as its my next project. D Dave Bour Desktop Solution Center 905.381.0077 [EMAIL PROTECTED] For those who just want it to work... Giving you complete IT peace of mind. (Sent via Blackberry - hence message may be shorter than my usual verbose responses) PIN 4cc364db (as of March 24, 2007) - Original Message - From: [EMAIL PROTECTED] [EMAIL PROTECTED] To: 'Asterisk-Users' asterisk-users@lists.digium.com Sent: Mon Jun 04 05:16:01 2007 Subject: [asterisk-users] Wireless IP Phone with external Telephone Book Hi, we are searching for wireless IP Phones (DECT preferred) with have an solution for an external telephone book. We don't want to enter all of our numbers into every telephone, but have one location for all the numbers and every phone looks them up there, e.g. an ldap server. We have tried Kirk but they are working on an solution without any information when it will be available. Does anyone knows an vendor which supports this feature ? Regards, Tobias Wolf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] zaptel on CENTOS servercd
I use CentOS4.4. successfully. I do something which is very odd for a Linux admin. I do a install everything. There is/was a reason for this. I was in a hurry to get a system online and didn't have time for a research project. I wrote a simple shell script to compile the apps (zaptel, libpri, asterisk),copy pre-defined .confs to asterisk -udev zaptel etc., set up vsftpd for provisioning up to 20 Polycom phones, set up DHCPD and finally the script removes a list of un-needed services left over from install everything The list of un-needed services continues to grow as I learn more. I know you won't find this in anyone's Best Practices wiki but I've used this process (install the os, run the script) 6 times for six different systems. It works flawlessly everytime. Takes an average of 51 minutes to get a complete system up and running. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tzafrir Cohen Sent: Monday, June 04, 2007 2:35 AM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] zaptel on CENTOS servercd One comment if I may: http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos does need some cleanup. Let alone merging of the two separate procedures and an update. Also: On Sun, Jun 03, 2007 at 06:22:57PM -0400, Lee Jenkins wrote: Khaled Chehab wrote: I suffered a lot from installing zaptel 1.4.2 on centos servercd ,Sure there is someone did that, Please in need to have the installation procedure step by step. Its too urgent for me . Thanks alot These scripts are working pretty well for me. They are based on a script I found on the wiki. # Script to download pre reqs for installation # == #!/bin/bash rpm --import http://mirror.centos.org/centos/RPM-GPG-KEY-CentOS-4 Isn't this done by the installer? yum -y install kernel-source bison openssl-devel Why do you need kernel-source? I know it is fun do download some extra 40Gigs, but apart from that, the kernel-devel / kernel-smp-devel is all you need to build zaptel. (likewise linux-headers-`uname -r` on Debian/Ubuntu ) yum -y update # Use kernel-devel instead of kernel-smp-devel # if not a SMP machine # |==| | | yum -y install gcc kernel kernel-smp-devel bison openssl-devel Hmmm you install kernel, but kernel-*smp*-devel . Strange ocmbination. Why not simply 'yum update' to get the latest kernel, if needed? This will upgrade either 'kernel' or 'kernel-smp', depending on the installed package. If you don't need a kernel upgrade you don't need the following: shutdown -r now # end script # Script to download asterisk, etc and build/install # == #!/bin/bash rm -f /usr/lib/asterisk/modules/* cd /usr/src rm -rf asterisk wget http://ftp.digium.com/pub/asterisk/releases/asterisk-1.2.18.tar.gz tar -zxvf asterisk-1.2.18.tar.gz mv asterisk-1.2.18 asterisk cd /usr/src rm -rf zaptel wget http://ftp.digium.com/pub/zaptel/releases/zaptel-1.2.17.1.tar.gz tar -zxvf zaptel-1.2.17.1.tar.gz mv zaptel-1.2.17.1 zaptel cd /usr/src rm -rf asterisk-addons wget http://ftp.digium.com/pub/asterisk/releases/asterisk-addons-1.2.6.tar.gz tar -zxvf asterisk-addons-1.2.6.tar.gz mv asterisk-addons-1.2.6 asterisk-addons cd /usr/src rm -rf asterisk-sounds wget http://ftp.digium.com/pub/asterisk/releases/asterisk-sounds-1.2.1.tar.gz tar -zxvf asterisk-sounds-1.2.1.tar.gz mv asterisk-sounds-1.2.1 asterisk-sounds cd /usr/src/zaptel make clean make install make config cd /usr/src/asterisk make clean make install make samples make config cd /usr/src/asterisk-addons make clean make install cd /usr/src/asterisk-sounds make install -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
Right now I can only speak to the WIP300 but I've been evaluating it for about a week now and really I have to say I'm fairly pleased. It works it works //well// but that's not to say it's perfect. - Physically the phone feels very light and cheap, that if you were to drop it that it might not survive very many of them. The buttons feel more like a toy than anything else but once you get beyond that it works. - Address book storage is ok the interface from the phone is fairly standard for what you would see in a cell phone and adding entries isn't really all that horrid of a task. You can also add entries via the web interface which does make for an easier way to add several entries but the lack of anything resembling a 'sync' function could be considered bothersome. - Call quality, so far so good however, I do believe the unit has a bit of an over touchy MIC.. the quality is clear but but it seems to pick up background noise and white noise pretty good. That's not to say that it will drown out your voice but you will hear the background if your in a server or climate controlled room. - The CPU on the phone does appear to be a bit underpowered. Two devices right next to each-other one a PC soft-phone and one the WIP 300 using the 'qualify' feature in asterisk you can clearly see a different in latency and how long it takes for the wip300 to process some sip transactions. This doesn't effect call quality but it is something worth noting if I'm taking the time to write this out. - Firmware: when you get the phone if it's running anything 1.0.9 upgrade to the latest from linksys, there are a slew of bugs that existing the factory shipping version that will likely make you think you really got cheated if you don't upgrade the firmware. Although make a note of the earlier thread upgrading has some bugs too, and don't try it form a mac. - I haven't tried the email function, as lack of intelligent keying (adaptive text for word completion) to me makes this a worthless feature. - Wireless, actually after changing to the latest version I've been fairly happy with the range coverage and life of the unit. You can load in multiple profiles for which AP you are talking to and the phone will register with that profile. You can associate different AP's with different SIP accounts which could be handy for traveling offices. The documentation doesn't mention it but you can create a profile that's a wild-card which will cause the phone to register to any open AP it finds. This I've found works fairly well as well as I can go from the east side to the west side of the building and the unit will switch AP's without much trouble, but do expect a dropped call in the process. If you have a mesh setup then the drop shouldn't happen but that's another story all together. I haven't played with the various encryption options so far as I've only been evaluating based on open access points. My thought is that likely the encryption may show more with the CPU load depending on how the unit manages this with it's chip set but even at that I don't think it will cause any red alarms. I could be wrong though. Last thing, one neat thing about the wip300 if you are adventurous is the fact that the firmware is under GPL... so if you really felt like it you could probably change the behavior of the phone. Anyway sorry for the long message but I felt like chiming in on this. All in all I don't think it's a horrible phone I do however think it's over priced for what it is but not enough demand on this type of device is always going to keep the price up in the air. -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX2 Trunk Problem
Hi I've two boxes connected over IAX2 trunk but suddenly my cli is getting flood with these messages: iax2_trunk_queue: Maximum data space exceeded and b'coz of that my agents are not able to hear any thing. when this happened that time there were 9 calls. my * version is 1.2.18 and 1.2.14 thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] G729 License
HI I bought 20 license from Digium and install in my server and b'coz of some problem I've to change my server is it possible that I can use those lice and register again in my new server ? Is it possible that I'll be able to use those lice in my old box also ? thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
On Jun 4, 2007, at 8:31 AM, Andrew Kohlsmith wrote: On Monday 04 June 2007 8:24 am, Bryan Laird wrote: - Physically the phone feels very light and cheap, that if you were to drop it that it might not survive very many of them. The buttons feel more like a toy than anything else but once you get beyond that it works. How are they for big hands? I'll have to do some checking around to see if I can find a rubber case for it or something, it's all concrete floors here. Considering I too have the sausage finger problem... the buttons are incredibly similar to what you find on the Nokia candy bar style phones. - Address book storage is ok the interface from the phone is fairly standard for what you would see in a cell phone and adding entries isn't really all that horrid of a task. You can also add entries via the web interface which does make for an easier way to add several entries but the lack of anything resembling a 'sync' function could be considered bothersome. Bugger. Last thing, one neat thing about the wip300 if you are adventurous is the fact that the firmware is under GPL... so if you really felt like it you could probably change the behavior of the phone. This I was not aware of. I will certainly evaluate this phone and it's bigger brother. Anyway sorry for the long message but I felt like chiming in on this. All in all I don't think it's a horrible phone I do however think it's over priced for what it is but not enough demand on this type of device is always going to keep the price up in the air. Your message is *exactly* the kind of reply I was hoping to get. Thank you so much for taking the time to write such a long response. I truly appreciate it. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] debug logs
Hi iam keep getting this log in my asterisk log is this harm anything, and how can stop this, any suggestions Jun 4 18:21:47 DEBUG[2093] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:49 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:49 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:50 DEBUG[2173] manager.c: Manager received command 'Command' ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 License
you can register twice after that you'll have to explain the reasons of changes to Digium Olivier Arun Kumar a crit: HI I bought 20 license from Digium and install in my server and b'coz of some problem I've to change my server is it possible that I can use those lice and register again in my new server ? Is it possible that I'll be able to use those lice in my old box also ? thanks arun ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel on CENTOS servercd
Jim Suber wrote: I use CentOS4.4. successfully. I do something which is very odd for a Linux admin. I do a install everything. There is/was a reason for this. I was in a hurry to get a system online and didn't have time for a research project. I wrote a simple shell script to compile the apps (zaptel, libpri, asterisk),copy pre-defined .confs to asterisk -udev zaptel etc., set up vsftpd for provisioning up to 20 Polycom phones, set up DHCPD and finally the script removes a list of un-needed services left over from install everything The list of un-needed services continues to grow as I learn more. I know you won't find this in anyone's Best Practices wiki but I've used this process (install the os, run the script) 6 times for six different systems. It works flawlessly everytime. Takes an average of 51 minutes to get a complete system up and running. As a Linux novice, I certainly agree. I posted such some time ago and was castigated for such blasphemy, but as you have discovered, this is certainly a quick way to get up and running. Use the 3 CD set rather than the single server CD and it goes MUCH quicker. Even quicker is to clone the hard drive then go change the few items that need changing. I have found that works well. Install everything then turn off services one doesn't need works well for both CentOs 3.x and 4.x, though Asterisk 1.4 seems to NEED the version 4. Any experience with CentOS 5?? Of course there are always the ( fill in the blank ) OS bigots that insist on using their favorite, and end up with problems. There are those who Linux is simply a means to an end, and not a religion! John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 License
On 6/4/07, Arun Kumar [EMAIL PROTECTED] wrote: HI I bought 20 license from Digium and install in my server and b'coz of some problem I've to change my server is it possible that I can use those lice and register again in my new server ? Is it possible that I'll be able to use those lice in my old box also ? Hi no its bound to ethernet address, when you change ethernet you need to register again with support of digium its only use for one Server ram thanks arun ___ --Bandwidth and Colocation provided by Easynews.com http://easynews.com/-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CDR timing
On Thu, 2007-05-31 at 15:44 -0500, Rob Schall wrote: A simple question but one I can't seem to find easily... I have 90 or so DIDs. For all outbound calls, I edit the callerid so that it will always read out main line's number. This poses a problem though, because the CDR detail isn't written until after everything is done. So when you look at the detail, the number you are calling is over written with this new caller id for some reason. Any thoughts how to better get an accurate reading of the calls, or another way to mask the caller id? Rob Rob-- You've left out some details, like where the callerid info you want to see is stored. Every channel keeps track of ani, rdnis, and dnid as well, and depending on the driver, usually only the ani or the plain callerid name/num end up in the CDR records. In the dialplan, you can attach other info onto a CDR by setting, for example, the CDR userfield, to one of these values, and hopefully, you can get this info out into a CDR so you can arrange your billing info accordingly... Also, keep in mind that the CDR is a separate data structure, and only gets updated at certain times, and you are only allowed to set certain fields via dialplan funcs. (and the CID info isn't on that list!). murf -- Steve Murphy Software Developer Digium smime.p7s Description: S/MIME cryptographic signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel on CENTOS servercd
Tzafrir Cohen wrote: One comment if I may: http://www.voip-info.org/wiki/view/Asterisk+Linux+Centos does need some cleanup. Let alone merging of the two separate procedures and an update. Also: On Sun, Jun 03, 2007 at 06:22:57PM -0400, Lee Jenkins wrote: Khaled Chehab wrote: I suffered a lot from installing zaptel 1.4.2 on centos servercd ,Sure there is someone did that, Please in need to have the installation procedure step by step. Its too urgent for me . Thanks alot These scripts are working pretty well for me. They are based on a script I found on the wiki. # Script to download pre reqs for installation # == #!/bin/bash rpm --import http://mirror.centos.org/centos/RPM-GPG-KEY-CentOS-4 Isn't this done by the installer? I don't think so because the scripts (yum) would fail without it. I'm using Centos 4.4 final if it makes a difference. yum -y install kernel-source bison openssl-devel Why do you need kernel-source? I know it is fun do download some extra 40Gigs, but apart from that, the kernel-devel / kernel-smp-devel is all you need to build zaptel. (likewise linux-headers-`uname -r` on Debian/Ubuntu ) Frankly, I am just following a script that I found on the wiki and I'm pretty new to linux still. http://www.voip-info.org/wiki/index.php?page=Asterisk+Linux+Centos yum -y update # Use kernel-devel instead of kernel-smp-devel # if not a SMP machine # |==| | | yum -y install gcc kernel kernel-smp-devel bison openssl-devel Hmmm you install kernel, but kernel-*smp*-devel . Strange ocmbination. Why not simply 'yum update' to get the latest kernel, if needed? This will upgrade either 'kernel' or 'kernel-smp', depending on the installed package. If you don't need a kernel upgrade you don't need the following: shutdown -r now # end script Thanks for the critique, Tzafrir. I will make these changes and try them out. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loud noise instead of MOH
Gang Chen wrote: - Original Message - From: Mauro Zanin [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, June 03, 2007 12:32 PM Subject: [asterisk-users] Loud noise instead of MOH Hi Everybody, I'm experiencing this kind of issue. One ASTERISK 1.2.17 is connected to a Bristuffed HFC single ISDN channel card. Everything seems to work but sometimes the third party caller when listening to MOH listens some SSH! instead of MOH, this is not continuos, MOH plays ok for, say, 20 seconds then the sound and then another 30 seconds of good MOH. We have some SIP phones for extensions (Grandstream 496 ATA and One SPX2000). Sometimes also ATAs hung, not really got ATAs at all. Does Anybody have some hint? Ciao a tutti Mauro Use native format .wav to play your moh instead of .mp3 may solve your problem. ___ This page may be of interest: http://astrecipes.net/?n=152 -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
Gordon Henderson wrote: On Sun, 3 Jun 2007, Andrew Kohlsmith wrote: On Sunday 03 June 2007 4:30 pm, Alex Crow wrote: No frills, specs look good, price seems excellent! http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519 That's a terrible phone. I've tried them. the screen is pretty much useless, the buttons are *TINY*, the battery life horrible, and the ringtones gimmicky. I have to disagree on at least one point here - Battery life. I don't think 3 or 4 days standby and several hours of talk time makes for a horrible battery life. The F1000G has other faillings, but battery life isn't one of them! If you compare this battery life to a decent DECT phone, it's still miserable. I'm used to these dect phones : http://www.bang-olufsen.com/UserFiles/File/Products/Technical%20Specifications/BeoCom6000_en.pdf I haven't tried WEP or WPA on these things, but the phones I've gotten rid of long ago due to their problems. Mine works with both WEP and WPA. It just occasionally won't re-attach to an access point (iwthout rebooting) and won't roam between access points at all. I think for the cost (and you can get them cheaper than from Scan!) they're not bad, but still in the experimental/toy category than something I'd deploy to clients. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Wireless IP Phone with external Telephone Book
On Mon, 4 Jun 2007, Tobias Wolf wrote:
Hi,
we are searching for wireless IP Phones (DECT preferred) with have an
solution for an external telephone book. We don't want to enter all of
our numbers into every telephone, but have one location for all the
numbers and every phone looks them up there, e.g. an ldap server.
We have tried Kirk but they are working on an solution without any
information when it will be available.
Does anyone knows an vendor which supports this feature ?
Since you're using them with asterisk (this is the asterisk-users list
after-all :), why not implement it in asterisk? Asterisk even has the
voice prompts already for it too, so someone's obviously though about it
in the past...
And this is exactly what I do in my systems, and the beauty is that it's
completely phone independant. You could have DECT phones, desk phones, WiFi
phones, soft phones, etc. all sharing the same phone-book.
Sure, it's not as neat as an in-phone solution (you can't easilly scroll
through the list on the phone), but it's universal over all phone types.
I implement a per-extension phone book, and a system phone book which
everyone has. And with a web interface, you can associate a name to the
number, then incoming caller-id works too, as long as the phone has a good
display.
Heres a working example from some old code I wrote - which I'm sure I got
off a web page (or the book) once upon a time in the past. (The list
commands here won't work with Siemens SIP/DECT phones as they won't let
you dial a trailling star, it's a # in the latest incarnation) This is
only 10 numbers per phone, but it's easy to change for more (eg. *000
through *099 for personal ones, or *100 through *199 for system ones -
left as an excercise to the reader ;-)
Gordon
;==
; Personal speed-dial:
; Let the user assign a personal speed dial code
; From *00 through *09
;==
;Setup: *0 then a single digit 0-9 then the phone number to store at that
location
exten = _*0X.,1,Answer()
exten = _*0X.,n,Set(me=${CALLERID(num)})
exten = _*0X.,n,Set(pos=${EXTEN:2:1})
exten = _*0X.,n,Set(num=${EXTEN:3})
exten = _*0X.,n,Set(DB(${me}/sd${pos})=${num})
exten = _*0X.,n,SayDigits(${num})
exten = _*0X.,n,playback(at)
exten = _*0X.,n,playback(position)
exten = _*0X.,n,SendText(*0${pos}=${num})
exten = _*0X.,n,SayDigits(${pos})
exten = _*0X.,n,Macro(starAck)
;List: *0X*
exten = _*0X*,1,Answer()
exten = _*0X*,n,Set(me=${CALLERID(num)})
exten = _*0X*,n,Set(pos=${EXTEN:2:1})
exten = _*0X*,n,Set(num=${DB(${me}/sd${pos})})
exten = _*0X*,n,GotoIf(${num}?:noNum)
exten = _*0X*,n,SendText(*0${pos}=${num})
exten = _*0X*,n,SayDigits(${num})
exten = _*0X*,n,playback(at)
exten = _*0X*,n,playback(position)
exten = _*0X*,n,SayDigits(0${pos})
exten = _*0X*,n,Hangup()
exten = _*0X*,n(noNum),playback(that-number)
exten = _*0X*,n,playback(is-not-in-the)
exten = _*0X*,n,playback(speed-dial)
exten = _*0X*,n,playback(system)
exten = _*0X*,n,Hangup()
;Recall and dial: *0X
exten = _*0X,1,Answer()
exten = _*0X,n,Set(me=${CALLERID(num)})
exten = _*0X,n,Set(pos=${EXTEN:2:1})
exten = _*0X,n,Set(num=${DB(${me}/sd${pos})})
exten = _*0X,n,GotoIf(${num}?:noNum)
exten = _*0X,n,SendText(Calling: ${num})
exten = _*0X,n,Goto(${num},1)
exten = _*0X,n(noNum),playback(that-number)
exten = _*0X,n,playback(is-not-in-the)
exten = _*0X,n,playback(speed-dial)
exten = _*0X,n,playback(system)
exten = _*0X,n,Hangup()
Gordon
___
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Re: [asterisk-users] CDR timing
Our call detail is located in 2 places. The master.csv file and in a mysql database. All outbound calls have: exten = _91NXXNXX,1,Set(CALLERID(all)=000-000-) (the zeros being our number) Right after that, it moves to the Dial command. The problem we are seeing, is in both the csv file and the mysql database, the src and clid are set to our main line's number. I need to to read the correct src. I don't care what the clid says. Rob Steve Murphy wrote: On Thu, 2007-05-31 at 15:44 -0500, Rob Schall wrote: A simple question but one I can't seem to find easily... I have 90 or so DIDs. For all outbound calls, I edit the callerid so that it will always read out main line's number. This poses a problem though, because the CDR detail isn't written until after everything is done. So when you look at the detail, the number you are calling is over written with this new caller id for some reason. Any thoughts how to better get an accurate reading of the calls, or another way to mask the caller id? Rob Rob-- You've left out some details, like where the callerid info you want to see is stored. Every channel keeps track of ani, rdnis, and dnid as well, and depending on the driver, usually only the ani or the plain callerid name/num end up in the CDR records. In the dialplan, you can attach other info onto a CDR by setting, for example, the CDR userfield, to one of these values, and hopefully, you can get this info out into a CDR so you can arrange your billing info accordingly... Also, keep in mind that the CDR is a separate data structure, and only gets updated at certain times, and you are only allowed to set certain fields via dialplan funcs. (and the CID info isn't on that list!). murf ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
Hi, I've tested several wifi phone (UtStarcom, Hitachi 3000 and 5000, and one Siemens). The Siemens is the best one, for a really cheaper price than hitachi. And was the only one which roams well between AP (same SSID, same channel) with WPA. Battery is still a problem, especially if the coverture is not very good everywhere. But that was the best one I could test... The reference is : Gigaset SL75 WLAN. Hope it helps Alban Le Lundi 4 Juin 2007 14:53, Bryan Laird a écrit : On Jun 4, 2007, at 8:31 AM, Andrew Kohlsmith wrote: On Monday 04 June 2007 8:24 am, Bryan Laird wrote: - Physically the phone feels very light and cheap, that if you were to drop it that it might not survive very many of them. The buttons feel more like a toy than anything else but once you get beyond that it works. How are they for big hands? I'll have to do some checking around to see if I can find a rubber case for it or something, it's all concrete floors here. Considering I too have the sausage finger problem... the buttons are incredibly similar to what you find on the Nokia candy bar style phones. - Address book storage is ok the interface from the phone is fairly standard for what you would see in a cell phone and adding entries isn't really all that horrid of a task. You can also add entries via the web interface which does make for an easier way to add several entries but the lack of anything resembling a 'sync' function could be considered bothersome. Bugger. Last thing, one neat thing about the wip300 if you are adventurous is the fact that the firmware is under GPL... so if you really felt like it you could probably change the behavior of the phone. This I was not aware of. I will certainly evaluate this phone and it's bigger brother. Anyway sorry for the long message but I felt like chiming in on this. All in all I don't think it's a horrible phone I do however think it's over priced for what it is but not enough demand on this type of device is always going to keep the price up in the air. Your message is *exactly* the kind of reply I was hoping to get. Thank you so much for taking the time to write such a long response. I truly appreciate it. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] addqueuemember recording and reporting problems
I am having a difficult time with the transition from agentcallback login... Here are a few of the isssues, I am logging in using chan_ local ie:local/8000 as the extension Call Detail records no longer show agent/ as the dstchannel show agents no longer shows the channels state show queues does not show the member Can anybody help? I have a ton of time invested in applications I developed that rely on this. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Re: click to call
Steven Have you been able to custommized the interface for babar's iax solution? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steven Sent: Lunes, 04 de Junio de 2007 07:17 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] Re: click to call I am using the free http://www.babarnazmi.citril.com/forum/viewtopic.php?t=7sid=fd8047cffb13074 969d3418064f4eb31 It is working as you described. It appears to be working well. -- -- Steven http://www.glimasoutheast.org Anton Krall [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] The idea is to put some kind of embedded app on the website so customers with mics can just click an icon or image and connect to our sales people or customer support staff... So far for what I've seen, there is some misconception of the terms.. click to dial can mean if you see a number on a webpage, click on it and your softphone will dial it.. but can also mean click on the image and it will connect you to the sales people, for example. I'm looking for the latter. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of mail-lists Sent: Jueves, 31 de Mayo de 2007 10:18 a.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] click to call Anton Krall wrote: I have been looking around for examples or code on making a click to call application for web sites... has anybody had any luck on this topic? Is there any open source code out ther that could do this? What we have done in the past is created url's like this : sip:4044565941. Xlite will register itself as the sip handler on your system. If you want a generic click to call (ability to call numbers on any given website) check out moziax ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] yum om centos
A lot of dependencies required for each module, I don't know the sequence of the rpms.Any way to know that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder Singh Sent: Monday, June 04, 2007 4:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] yum om centos independently install each rpm via rpm command :-/ On 04/06/07, Khaled Chehab [EMAIL PROTECTED] wrote: I have 2 servers, one connected to internet and the other is on a private lan have no access to internet. On the first server I update the kernel by yum update And installed asterisk prerequisite module yum install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf \ libtool make automake automake14 automake15 automake16 automake17 \ bison byacc flex libtermcap libtermcap-devel newt newt-devel \ ncurses ncurses-devel openssl-devel zlib zlib-devel krb5-devel I zipped /var/cache/yum from the first server and extract it on the second server at the same directory. On the second server I tried to update using yum update but the yum update failed. How can I do that with out connecting the second server to internet . Khaled Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
Zoa wrote: Gordon Henderson wrote: On Sun, 3 Jun 2007, Andrew Kohlsmith wrote: On Sunday 03 June 2007 4:30 pm, Alex Crow wrote: No frills, specs look good, price seems excellent! http://www.scan.co.uk/Products/ProductInfo.asp?WebProductID=369519 That's a terrible phone. I've tried them. the screen is pretty much useless, the buttons are *TINY*, the battery life horrible, and the ringtones gimmicky. I have to disagree on at least one point here - Battery life. I don't think 3 or 4 days standby and several hours of talk time makes for a horrible battery life. The F1000G has other faillings, but battery life isn't one of them! If you compare this battery life to a decent DECT phone, it's still miserable. I'm used to these dect phones : http://www.bang-olufsen.com/UserFiles/File/Products/Technical%20Specifications/BeoCom6000_en.pdf [snip] It's not really a fair comparison though, DECT was designed from the offset to be used with portable phones on low-power batteries. Wifi was never designed for this so it's comparatively power hungry. The F1000G in my opinion is still pretty good battery life for a wifi phone, I've seen some that will not last a day in standby. Looking at the OP's requirements list in the first post, there is nothing currently on the market which will cover anything like all those features (and do it well!). I've currently got several Nokia Wifi/GSM phones sat on my desk, they are difficult to configure and very quirky, frankly not even good enough to be considered a techie's toy. I am told 3rd party softphone clients such as TruPhone work a lot better than the built-in SIP client but I'm yet to test any of these. The main problem is they have a habit of constantly losing connection with my access points. Even the F1000G and F3000 phones I have here don't do that. I'm yet to be convinced that wifi in it's current state is any use for telephony at all. DECT works so much better, it just needs someone to make a fully functioning SIP DECT phone. The Siemens is good but they need to work on more SIP functions, although proper transfers should be possible soon. I also have some Philips DECT SIP equipment next to my desk to look at when I've got a chance! cheers, Paul. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] FX Dialing Odd
Here's a possible bug, or more likely, I'm just missing something.
We have a pots card in one of our asterisk boxes. Its a simple asterisk
setup with one FXO/FXS card and basic static extensions file, etc. When
we dial out over the pots line, 4 out of 5 times, it will work. However,
every 4 or 5 times, we get an error back from the provider that says
The number you have dialed. blah blah blah. The interesting there
there, is that the number it quotes us, adds our local area code in
front of it. And this only seems to happen on long distance context
numbers. So when we dial 630-XXX-, sometimes we are sent to
312-681-XXX-.
Here's our configs:
zaptel.conf
loadzone=us
defaultzone=us
fxoks=1-2
fxsks=3
zapata.conf
[trunkgroups]
[channels]
language=en
context=internal
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
callgroup=1
pickupgroup=1
immediate=no
;define channels
context=internal
signalling=fxo_ks
channel = 1-2
context=zapchans
signalling=fxs_ks
group=1
channel = 3
extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no priorityjumping=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;TRUNK=Zap/g1 ; Trunk interface
TRUNK=Zap/3 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
[trunkint]
exten = _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten = _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
exten = _91NXXNXX,1,Verbose(LD)
exten = _91NXXNXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,wW)
exten = _91NXXNXX,n,Hangup()
[trunklocal]
exten = _9NXX,1,Verbose(LOCAL)
exten = _9NXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,wW)
exten = _9NXX,n,Hangup()
[trunktollfree]
exten = _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[outgoing]
include = trunklocal
include = trunktollfree
include = trunkld
;ZAP Channels
[zapchans]
exten = 3,1,Dial(ZAP/1-1)
exten = 3,2,Hangup()
exten = 4,1,Dial(ZAP/2-1)
exten = 4,2,Hangup()
exten = s,1,Answer()
exten = s,2,NoOp(${CHANNEL:4:1})
exten = s,3,Goto(${CHANNEL:4:1},1)
exten = s,4,Hangup()
[incoming]
include =internal
[internal]
include = outgoing
exten = ,1,Dial(Zap/3/1630XXX,,wW)
exten = ,2,Hangup()
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[asterisk-users] background dialing
Hello. Is it possible to dial in background 2 or more different numbers while the same uninterrupted soundfile is playing? Something like this: exten = Answer exten = Playback (hello-bla-bla-we are trying to connect you-play-music) exten = Dial (SIP110/15 and after 15s DIAL SIP111 without interrupt the Playback-Music) Is that possible? Thank you in advance. regards t. -- knowledgeTools® ... managing complexity. -- knowledgeTools International GmbH Wallstraße 15 / 15 a 10179 Berlin Fon: +49 30 726 169 20 Fax: +49 30 726 169 249 [EMAIL PROTECTED] www.knowledgetools.de Sitz Berlin, AG Berlin-Charlottenburg, HRB 86378 Geschäftsführer: Oliver Seyboldt, Reinhard Kunz -- This eMail communication (and any attachment/s) may contain confidential or privileged information and is intended only for the individual(s) or entity named above and to others who have been specifically authorized to receive it. If you are not the intended recipient, please do not read, copy, use or disclose the contents of this communication to others. Please notify the sender that you have received this e-mail in error by reply e-mail, and delete the e-mail subsequently. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any codec passthru mode
Asterisk by default uses the codec preferred by other device/client . Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough to check if it can avoid transcoding by forcing same codec on other side of conversation . If both sides prefer g729 then asterisk does not do transcoding but if one side prefer gsm and other prefers g729 and the gsm side can also support g729 still asterisk will transcode . Someone posted a patch to this in mantis bug tracking system at digium for 1.2 .. google for it and maybe you can find . On 31/05/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Does anybody has any documentation on codec negotiation within asterisk? Well im using free g729 codec for testing purposes. i mentioned g729 just as an example. whatever codec is mentioned in user perefernce, asterisk uses ulaw to throw out the call. On 5/30/07, Marco Mouta [EMAIL PROTECTED] wrote: so you r sure you have g729 licences installed and ur * is transcoding your RTP streaming? Test the work flow with disallow=all and allow=g729, can be my mistake but I remember to read somewhere on the net any issue about codec negotiating precedence when you use allow=all. good luck On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, My configuration is: USER (connects to) ASTERISK---(connects to)---CARRIER-OUT i want the user preffered codec to pass thru asterisk to carrier-out. what i mean is: USER (user uses g729) ASTERISK---(asterisk should use g729 for dialing out)---CARRIER-OUT instead, this is what happens USER (user uses g729) ASTERISK---(asterisk uses g711u)---CARRIER-OUT How can i force asterisk to use user preffered codec for dialing out so that my asterisk machine saves time by no conversion USER PREFERENCE IS disallow=all allow=g729 CARRIER PREFERENCE IS allow=all Anybody who can help? -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FX Dialing Odd
Rob Schall wrote: Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However,every 4 or 5 times, we get an error back from the provider that says The number you have dialed. blah blah blah. Common defect in the Zaptel driver. It does NOT listen for dial tone, so if you have not inserted a w or three into the dial string, it will dial before the Central Office is ready, and it may miss a digit, causing misdials. Curious that cheap modems years ago learned to listen for dial tone, but the Zaptel driver doesn't, and of course this is considered a feature request rather than a bug, and no one seems to want to fix it. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
On Monday 04 June 2007 10:28 am, Paul Hayes wrote: Looking at the OP's requirements list in the first post, there is nothing currently on the market which will cover anything like all those features (and do it well!). I've got the WIP300 and 330 on my list, with the latter being the more likely candidate, as I can throw up custom apps once I figure out how it's done. :-) I like the idea of a wifi phone running Linux though, so both of these options will have to be investigated. but I'm yet to test any of these. The main problem is they have a habit of constantly losing connection with my access points. Even the F1000G and F3000 phones I have here don't do that. My F1000G phones *CONSTANTLY* lost connection with my WRT54, and it had nothing to do with signal strength, as the access point was less than 10 feet away from my desk, with nothing between to interfere. :-( I'm yet to be convinced that wifi in it's current state is any use for telephony at all. DECT works so much better, it just needs someone to make a fully functioning SIP DECT phone. The Siemens is good but they need to work on more SIP functions, although proper transfers should be possible soon. I am also slowly coming to this conclusion. Polycom recently acquired SpectraLink, who've got many years in the wireless phone business. They've got both Wifi and DECT offerings, but nothing with bluetooth, so the search continues. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ringback detection
It just might be that your carrier is not sending ring . You can use 'r' in asterisk dial command in extensions.conf to generate ring from asterisk . On 31/05/07, dima [EMAIL PROTECTED] wrote: Hello, everyone. Could anyone explain me how does ringback detection works in asterisk. Sometimes, when making a call, my asterisk box doesn't detect a ringback and I just hear silence until the other party picks up the phone. I've checked the SIP messages and they are ok (I'm getting 183 session in progress), so I guess I should be debugging the RTP packets. From then on I'm stuck. Does anyone know what type of packets I should be looking for? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any codec passthru mode
That just seems really, REALLY dumb for a program of this magnitude. I know this has been patched here and there by one person or another, but does anyone know if any of these patches to make CODEC negotiation actually, you know, negotiate a CODEC will ever make it into the core src? Jaswinder Singh wrote: Asterisk by default uses the codec preferred by other device/client . Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough to check if it can avoid transcoding by forcing same codec on other side of conversation . If both sides prefer g729 then asterisk does not do transcoding but if one side prefer gsm and other prefers g729 and the gsm side can also support g729 still asterisk will transcode . Someone posted a patch to this in mantis bug tracking system at digium for 1.2 .. google for it and maybe you can find . On 31/05/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Does anybody has any documentation on codec negotiation within asterisk? Well im using free g729 codec for testing purposes. i mentioned g729 just as an example. whatever codec is mentioned in user perefernce, asterisk uses ulaw to throw out the call. On 5/30/07, Marco Mouta [EMAIL PROTECTED] wrote: so you r sure you have g729 licences installed and ur * is transcoding your RTP streaming? Test the work flow with disallow=all and allow=g729, can be my mistake but I remember to read somewhere on the net any issue about codec negotiating precedence when you use allow=all. good luck On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, My configuration is: USER (connects to) ASTERISK---(connects to)---CARRIER-OUT i want the user preffered codec to pass thru asterisk to carrier-out. what i mean is: USER (user uses g729) ASTERISK---(asterisk should use g729 for dialing out)---CARRIER-OUT instead, this is what happens USER (user uses g729) ASTERISK---(asterisk uses g711u)---CARRIER-OUT How can i force asterisk to use user preffered codec for dialing out so that my asterisk machine saves time by no conversion USER PREFERENCE IS disallow=all allow=g729 CARRIER PREFERENCE IS allow=all Anybody who can help? -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
Alban, Thanks! Where on earth did you source this? I can't seen to find hide nor hair of it here in the UK :( Alex On Mon, 2007-06-04 at 16:01 +0200, Alban wrote: Hi, I've tested several wifi phone (UtStarcom, Hitachi 3000 and 5000, and one Siemens). The Siemens is the best one, for a really cheaper price than hitachi. And was the only one which roams well between AP (same SSID, same channel) with WPA. Battery is still a problem, especially if the coverture is not very good everywhere. But that was the best one I could test... The reference is : Gigaset SL75 WLAN. Hope it helps Alban Le Lundi 4 Juin 2007 14:53, Bryan Laird a écrit : On Jun 4, 2007, at 8:31 AM, Andrew Kohlsmith wrote: On Monday 04 June 2007 8:24 am, Bryan Laird wrote: - Physically the phone feels very light and cheap, that if you were to drop it that it might not survive very many of them. The buttons feel more like a toy than anything else but once you get beyond that it works. How are they for big hands? I'll have to do some checking around to see if I can find a rubber case for it or something, it's all concrete floors here. Considering I too have the sausage finger problem... the buttons are incredibly similar to what you find on the Nokia candy bar style phones. - Address book storage is ok the interface from the phone is fairly standard for what you would see in a cell phone and adding entries isn't really all that horrid of a task. You can also add entries via the web interface which does make for an easier way to add several entries but the lack of anything resembling a 'sync' function could be considered bothersome. Bugger. Last thing, one neat thing about the wip300 if you are adventurous is the fact that the firmware is under GPL... so if you really felt like it you could probably change the behavior of the phone. This I was not aware of. I will certainly evaluate this phone and it's bigger brother. Anyway sorry for the long message but I felt like chiming in on this. All in all I don't think it's a horrible phone I do however think it's over priced for what it is but not enough demand on this type of device is always going to keep the price up in the air. Your message is *exactly* the kind of reply I was hoping to get. Thank you so much for taking the time to write such a long response. I truly appreciate it. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] debug logs
On Mon, Jun 04, 2007 at 06:34:37PM +0530, ram wrote: Hi iam keep getting this log in my asterisk log is this harm anything, and how can stop this, any suggestions Jun 4 18:21:47 DEBUG[2093] chan_sip.c: Stopping retransmission on '[EMAIL PROTECTED]' of Request 102: Match Found Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:48 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:49 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:49 DEBUG[2173] manager.c: Manager received command 'Command' Jun 4 18:21:50 DEBUG[2173] manager.c: Manager received command 'Command' Those are debug logs. They are supposed to be full of debug messages. No point in getting excited over them. Specifically the manager interface command Command allows running CLI commands. This notifies you that it has been used (IIRC). -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FX Dialing Odd
But if this was the case, then why would the message playback (from the provider) read back the digits from the start. I mean, I dialed 630-XXX-XXX and it reads back 312 (my local area code)-630-XXX-X I would think if it wasn't waiting, then it would do like a 312-30X-XX-XX Rob John Novack wrote: Rob Schall wrote: Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However,every 4 or 5 times, we get an error back from the provider that says The number you have dialed. blah blah blah. Common defect in the Zaptel driver. It does NOT listen for dial tone, so if you have not inserted a w or three into the dial string, it will dial before the Central Office is ready, and it may miss a digit, causing misdials. Curious that cheap modems years ago learned to listen for dial tone, but the Zaptel driver doesn't, and of course this is considered a feature request rather than a bug, and no one seems to want to fix it. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no dtmf pcom 650 only outbound calls
PROBLEM...NO DTMF ON OUTBOUND CALLS 1 ASTERISK FORWARDS THE DIGITS Got rfc2833 RTP packet from 66.108.217.191:2256 (type 101, seq 279, ts -1975142833, len 4, mark 0, event 0009, end 1, duration 1600) Sent RTP packet to 209.212.88.161:32872 (type 101, seq 63407, ts 60776, len 4) Sent RTP packet to 209.212.88.161:32872 (type 101, seq 63408, ts 60776, len 4) Sent RTP packet to 209.212.88.161:32872 (type 101, seq 63409, ts 60776, len 4) Sent RTP packet to 209.212.88.161:32872 (type 101, seq 63410, ts 60776, len 4) Sent RTP packet to 209.212.88.161:32872 (type 101, seq 63410, ts 60776, len 4) Sent RTP packet to 209.212.88.161:32872 (type 101, seq 63410, ts 60776, len 4) 2 POLYCOM IS SET CORRECTLY DTMF tone.dtmf.level=-15 tone.dtmf.onTime=50 tone.dtmf.offTime=50 tone.dtmf.chassis.masking=0 tone.dtmf.stim.pac.offHookOnly=0 tone.dtmf.viaRtp=1 tone.dtmf.rfc2833Control=1 tone.dtmf.rfc2833Payload=101/ chord_sets 3 DTMF WORKS FINE WHEN CHECKING VOICEMAIL 4 WHEN CHANGE CARRIERS NO CHANGE 5 CANNOT HEAR DTMF IN EAR WHEN CALL A PHONE 6 only with pcom650 never a problem with 601 and 501 7 when change to inband dtmf (in freepbx and in pcom config) dtmf works but its spotty not acceptable ASTERISK VER 1.12.13 thank you in advance _ Get a preview of Live Earth, the hottest event this summer - only on MSN http://liveearth.msn.com?source=msntaglineliveearthhm ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any codec passthru mode
some work has been done here: http://bugs.digium.com/view.php?id=4825 but seems to be quite death and probably not directly applicable to current asterisk src :'( SIP wrote: That just seems really, REALLY dumb for a program of this magnitude. I know this has been patched here and there by one person or another, but does anyone know if any of these patches to make CODEC negotiation actually, you know, negotiate a CODEC will ever make it into the core src? Jaswinder Singh wrote: Asterisk by default uses the codec preferred by other device/client . Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough to check if it can avoid transcoding by forcing same codec on other side of conversation . If both sides prefer g729 then asterisk does not do transcoding but if one side prefer gsm and other prefers g729 and the gsm side can also support g729 still asterisk will transcode . Someone posted a patch to this in mantis bug tracking system at digium for 1.2 .. google for it and maybe you can find . On 31/05/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Does anybody has any documentation on codec negotiation within asterisk? Well im using free g729 codec for testing purposes. i mentioned g729 just as an example. whatever codec is mentioned in user perefernce, asterisk uses ulaw to throw out the call. On 5/30/07, Marco Mouta [EMAIL PROTECTED] wrote: so you r sure you have g729 licences installed and ur * is transcoding your RTP streaming? Test the work flow with disallow=all and allow=g729, can be my mistake but I remember to read somewhere on the net any issue about codec negotiating precedence when you use allow=all. good luck On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, My configuration is: USER (connects to) ASTERISK---(connects to)---CARRIER-OUT i want the user preffered codec to pass thru asterisk to carrier-out. what i mean is: USER (user uses g729) ASTERISK---(asterisk should use g729 for dialing out)---CARRIER-OUT instead, this is what happens USER (user uses g729) ASTERISK---(asterisk uses g711u)---CARRIER-OUT How can i force asterisk to use user preffered codec for dialing out so that my asterisk machine saves time by no conversion USER PREFERENCE IS disallow=all allow=g729 CARRIER PREFERENCE IS allow=all Anybody who can help? -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FX Dialing Odd
Don't overthink this Error messages from providers are frequently misleading and inaccurate Some only use one recording for anything they think they can't process. Add at least one w to the dial string and see if all the misdials don't go away. Also check the list archives for MANY such complaints John Novack Rob Schall wrote: But if this was the case, then why would the message playback (from the provider) read back the digits from the start. I mean, I dialed 630-XXX-XXX and it reads back 312 (my local area code)-630-XXX-X I would think if it wasn't waiting, then it would do like a 312-30X-XX-XX Rob John Novack wrote: Rob Schall wrote: Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However,every 4 or 5 times, we get an error back from the provider that says The number you have dialed. blah blah blah. Common defect in the Zaptel driver. It does NOT listen for dial tone, so if you have not inserted a w or three into the dial string, it will dial before the Central Office is ready, and it may miss a digit, causing misdials. Curious that cheap modems years ago learned to listen for dial tone, but the Zaptel driver doesn't, and of course this is considered a feature request rather than a bug, and no one seems to want to fix it. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] any codec passthru mode
has anybody made a patch for asterisk 1.4*? On 6/4/07, Jaswinder Singh [EMAIL PROTECTED] wrote: Asterisk by default uses the codec preferred by other device/client . Asterisk 1.2 ( dunno abt 1.4 specifically) is not intelligent enough to check if it can avoid transcoding by forcing same codec on other side of conversation . If both sides prefer g729 then asterisk does not do transcoding but if one side prefer gsm and other prefers g729 and the gsm side can also support g729 still asterisk will transcode . Someone posted a patch to this in mantis bug tracking system at digium for 1.2 .. google for it and maybe you can find . On 31/05/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Does anybody has any documentation on codec negotiation within asterisk? Well im using free g729 codec for testing purposes. i mentioned g729 just as an example. whatever codec is mentioned in user perefernce, asterisk uses ulaw to throw out the call. On 5/30/07, Marco Mouta [EMAIL PROTECTED] wrote: so you r sure you have g729 licences installed and ur * is transcoding your RTP streaming? Test the work flow with disallow=all and allow=g729, can be my mistake but I remember to read somewhere on the net any issue about codec negotiating precedence when you use allow=all. good luck On 5/30/07, Rizwan Hisham [EMAIL PROTECTED] wrote: Hi all, My configuration is: USER (connects to) ASTERISK---(connects to)---CARRIER-OUT i want the user preffered codec to pass thru asterisk to carrier-out. what i mean is: USER (user uses g729) ASTERISK---(asterisk should use g729 for dialing out)---CARRIER-OUT instead, this is what happens USER (user uses g729) ASTERISK---(asterisk uses g711u)---CARRIER-OUT How can i force asterisk to use user preffered codec for dialing out so that my asterisk machine saves time by no conversion USER PREFERENCE IS disallow=all allow=g729 CARRIER PREFERENCE IS allow=all Anybody who can help? -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Esta mensagem (incluindo quaisquer anexos) pode conter informação confidencial para uso exclusivo do destinatário. Se não for o destinatário pretendido, não deverá usar, distribuir ou copiar este e-mail. Se recebeu esta mensagem por engano, por favor informe o emissor e elimine-a imediatamente. Obrigado. This e-mail message is intended only for individual(s) to whom it is addressed and may contain information that is privileged, confidential, proprietary, or otherwise exempt from disclosure under applicable law. If you believe you have received this message in error, please advise the sender by return e-mail and delete it from your mailbox. Thank you. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Rizwan Hisham Software Engineer AXVOICE Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FX Dialing Odd
Also, in the dial command the w says its for the *1 recording. Not waiting. Is the documentation wrong? What is the correct way to wait in the dial command? Rob John Novack wrote: Rob Schall wrote: Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However,every 4 or 5 times, we get an error back from the provider that says The number you have dialed. blah blah blah. Common defect in the Zaptel driver. It does NOT listen for dial tone, so if you have not inserted a w or three into the dial string, it will dial before the Central Office is ready, and it may miss a digit, causing misdials. Curious that cheap modems years ago learned to listen for dial tone, but the Zaptel driver doesn't, and of course this is considered a feature request rather than a bug, and no one seems to want to fix it. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
We have it (in belgium) http://www.voipsolutions.be/phones/dect-sip-phones/siemens-gigaset-sl75-wlan-voip-phone.html I still think DECT is better though :) Zoa Alex Crow wrote: Alban, Thanks! Where on earth did you source this? I can't seen to find hide nor hair of it here in the UK :( Alex On Mon, 2007-06-04 at 16:01 +0200, Alban wrote: Hi, I've tested several wifi phone (UtStarcom, Hitachi 3000 and 5000, and one Siemens). The Siemens is the best one, for a really cheaper price than hitachi. And was the only one which roams well between AP (same SSID, same channel) with WPA. Battery is still a problem, especially if the coverture is not very good everywhere. But that was the best one I could test... The reference is : Gigaset SL75 WLAN. Hope it helps Alban Le Lundi 4 Juin 2007 14:53, Bryan Laird a écrit : On Jun 4, 2007, at 8:31 AM, Andrew Kohlsmith wrote: On Monday 04 June 2007 8:24 am, Bryan Laird wrote: - Physically the phone feels very light and cheap, that if you were to drop it that it might not survive very many of them. The buttons feel more like a toy than anything else but once you get beyond that it works. How are they for big hands? I'll have to do some checking around to see if I can find a rubber case for it or something, it's all concrete floors here. Considering I too have the sausage finger problem... the buttons are incredibly similar to what you find on the Nokia candy bar style phones. - Address book storage is ok the interface from the phone is fairly standard for what you would see in a cell phone and adding entries isn't really all that horrid of a task. You can also add entries via the web interface which does make for an easier way to add several entries but the lack of anything resembling a 'sync' function could be considered bothersome. Bugger. Last thing, one neat thing about the wip300 if you are adventurous is the fact that the firmware is under GPL... so if you really felt like it you could probably change the behavior of the phone. This I was not aware of. I will certainly evaluate this phone and it's bigger brother. Anyway sorry for the long message but I felt like chiming in on this. All in all I don't think it's a horrible phone I do however think it's over priced for what it is but not enough demand on this type of device is always going to keep the price up in the air. Your message is *exactly* the kind of reply I was hoping to get. Thank you so much for taking the time to write such a long response. I truly appreciate it. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] wifi sip phone real-world experiences?
Alex, I bought them at Beronet (shop.beronet.com), a german company. I'm located in France, no problem for them to send them here... Alban Le Lundi 4 Juin 2007 17:29, Alex Crow a écrit : Alban, Thanks! Where on earth did you source this? I can't seen to find hide nor hair of it here in the UK :( Alex On Mon, 2007-06-04 at 16:01 +0200, Alban wrote: Hi, I've tested several wifi phone (UtStarcom, Hitachi 3000 and 5000, and one Siemens). The Siemens is the best one, for a really cheaper price than hitachi. And was the only one which roams well between AP (same SSID, same channel) with WPA. Battery is still a problem, especially if the coverture is not very good everywhere. But that was the best one I could test... The reference is : Gigaset SL75 WLAN. Hope it helps Alban Le Lundi 4 Juin 2007 14:53, Bryan Laird a écrit : On Jun 4, 2007, at 8:31 AM, Andrew Kohlsmith wrote: On Monday 04 June 2007 8:24 am, Bryan Laird wrote: - Physically the phone feels very light and cheap, that if you were to drop it that it might not survive very many of them. The buttons feel more like a toy than anything else but once you get beyond that it works. How are they for big hands? I'll have to do some checking around to see if I can find a rubber case for it or something, it's all concrete floors here. Considering I too have the sausage finger problem... the buttons are incredibly similar to what you find on the Nokia candy bar style phones. - Address book storage is ok the interface from the phone is fairly standard for what you would see in a cell phone and adding entries isn't really all that horrid of a task. You can also add entries via the web interface which does make for an easier way to add several entries but the lack of anything resembling a 'sync' function could be considered bothersome. Bugger. Last thing, one neat thing about the wip300 if you are adventurous is the fact that the firmware is under GPL... so if you really felt like it you could probably change the behavior of the phone. This I was not aware of. I will certainly evaluate this phone and it's bigger brother. Anyway sorry for the long message but I felt like chiming in on this. All in all I don't think it's a horrible phone I do however think it's over priced for what it is but not enough demand on this type of device is always going to keep the price up in the air. Your message is *exactly* the kind of reply I was hoping to get. Thank you so much for taking the time to write such a long response. I truly appreciate it. -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=-=- Bryan Laird, Sr. Manager CM Operations -+- Cablemodems are the gateway to the Internet. The Internet is a gateway to some things that are better left un-seen. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FX Dialing Odd
Not option w
add a w in the DIAL STRING
Guess you didn't search the archives?
Sometime in 1.2 this feature was fixed to work with pulse dial as well.
example:
exten = s,1,Dial(ZAP/g4/w(${ARG1:3:4}),360,Tt)
John Novack
Rob Schall wrote:
Also, in the dial command the w says its for the *1 recording. Not
waiting. Is the documentation wrong? What is the correct way to wait in
the dial command?
Rob
John Novack wrote:
Rob Schall wrote:
Here's a possible bug, or more likely, I'm just missing something.
We have a pots card in one of our asterisk boxes. Its a simple
asterisk setup with one FXO/FXS card and basic static extensions
file, etc. When we dial out over the pots line, 4 out of 5 times, it
will work. However,every 4 or 5 times, we get an error back from the
provider that says The number you have dialed. blah blah blah.
Common defect in the Zaptel driver.
It does NOT listen for dial tone, so if you have not inserted a w or
three into the dial string, it will dial before the Central Office is
ready, and it may miss a digit, causing misdials.
Curious that cheap modems years ago learned to listen for dial tone,
but the Zaptel driver doesn't, and of course this is considered a
feature request rather than a bug, and no one seems to want to fix it.
John Novack
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Re: [asterisk-users] FX Dialing Odd
Asterisk does NOT wait for dialtone when going offhook on an FXO port.
Asterisk is sending digits to the telco too quickly.
Add a w (.5 second wait) before the extension to be dialed. Example:
exten = _91NXXNXX,n,Dial(${TRUNK}/w${EXTEN:${TRUNKMSD}},,wW)
Rob Schall wrote:
Here's a possible bug, or more likely, I'm just missing something.
We have a pots card in one of our asterisk boxes. Its a simple asterisk
setup with one FXO/FXS card and basic static extensions file, etc. When
we dial out over the pots line, 4 out of 5 times, it will work. However,
every 4 or 5 times, we get an error back from the provider that says
The number you have dialed. blah blah blah. The interesting there
there, is that the number it quotes us, adds our local area code in
front of it. And this only seems to happen on long distance context
numbers. So when we dial 630-XXX-, sometimes we are sent to
312-681-XXX-.
Here's our configs:
zaptel.conf
loadzone=us
defaultzone=us
fxoks=1-2
fxsks=3
zapata.conf
[trunkgroups]
[channels]
language=en
context=internal
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
callgroup=1
pickupgroup=1
immediate=no
;define channels
context=internal
signalling=fxo_ks
channel = 1-2
context=zapchans
signalling=fxs_ks
group=1
channel = 3
extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no priorityjumping=no
[globals]
CONSOLE=Console/dsp ; Console interface for demo
;TRUNK=Zap/g1 ; Trunk interface
TRUNK=Zap/3 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
[trunkint]
exten = _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten = _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
exten = _91NXXNXX,1,Verbose(LD)
exten = _91NXXNXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,wW)
exten = _91NXXNXX,n,Hangup()
[trunklocal]
exten = _9NXX,1,Verbose(LOCAL)
exten = _9NXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,wW)
exten = _9NXX,n,Hangup()
[trunktollfree]
exten = _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[outgoing]
include = trunklocal
include = trunktollfree
include = trunkld
;ZAP Channels
[zapchans]
exten = 3,1,Dial(ZAP/1-1)
exten = 3,2,Hangup()
exten = 4,1,Dial(ZAP/2-1)
exten = 4,2,Hangup()
exten = s,1,Answer()
exten = s,2,NoOp(${CHANNEL:4:1})
exten = s,3,Goto(${CHANNEL:4:1},1)
exten = s,4,Hangup()
[incoming]
include =internal
[internal]
include = outgoing
exten = ,1,Dial(Zap/3/1630XXX,,wW)
exten = ,2,Hangup()
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Re: [asterisk-users] wifi sip phone real-world experiences?
well, the best I got is the tc300/arcor/twintel a gsm/wifi from pirelli - http://www.pirellibroadband.com/en_IT/browser/attachments/pdf/DPL10.pdf tried many wifi phones, that's the best we got. Long lifetime for the battery, good reception, roaming between Access points with the same network... good enough for us and cheap. livier Andrew Kohlsmith a crit: On Monday 04 June 2007 10:28 am, Paul Hayes wrote: Looking at the OP's requirements list in the first post, there is nothing currently on the market which will cover anything like all those features (and do it well!). I've got the WIP300 and 330 on my list, with the latter being the more likely candidate, as I can throw up custom apps once I figure out how it's done. :-) I like the idea of a wifi phone running Linux though, so both of these options will have to be investigated. but I'm yet to test any of these. The main problem is they have a habit of constantly losing connection with my access points. Even the F1000G and F3000 phones I have here don't do that. My F1000G phones *CONSTANTLY* lost connection with my WRT54, and it had nothing to do with signal strength, as the access point was less than 10 feet away from my desk, with nothing between to interfere. :-( I'm yet to be convinced that wifi in it's current state is any use for telephony at all. DECT works so much better, it just needs someone to make a fully functioning SIP DECT phone. The Siemens is good but they need to work on more SIP functions, although proper transfers should be possible soon. I am also slowly coming to this conclusion. Polycom recently acquired SpectraLink, who've got many years in the wireless phone business. They've got both Wifi and DECT offerings, but nothing with bluetooth, so the search continues. :-) -A. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] AEL2 Includes in Macro...
Where's Steve Murphy when you need him? :-)
This doesn't seem to work in AEL2...
Macro foo(arg1) {
.
Includes {
Hangup;
}
}
The error is: File: /etc/asterisk/extensions.ael, Line 59, Cols: 5-12:
Error: syntax error, unexpected KW_INCLUDES, expecting '}'
The same error does not occur when the includes is in a context.
I need to have the ability to include my hangup routine in macros, as
theoretically, a hang up could occur while asterisk is processing code
from the macro.
This is Asterisk 1.4.4
Doug.
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Re: [asterisk-users] FX Dialing Odd
What happens if you connect the fxo to the fxs and try several attempts at completing a call? This should at least tell you if the issue is outdialed digit issues or telco receipt issues. Dave On Mon, 2007-06-04 at 10:30 -0500, Rob Schall wrote: But if this was the case, then why would the message playback (from the provider) read back the digits from the start. I mean, I dialed 630-XXX-XXX and it reads back 312 (my local area code)-630-XXX-X I would think if it wasn't waiting, then it would do like a 312-30X-XX-XX Rob John Novack wrote: Rob Schall wrote: Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However,every 4 or 5 times, we get an error back from the provider that says The number you have dialed. blah blah blah. Common defect in the Zaptel driver. It does NOT listen for dial tone, so if you have not inserted a w or three into the dial string, it will dial before the Central Office is ready, and it may miss a digit, causing misdials. Curious that cheap modems years ago learned to listen for dial tone, but the Zaptel driver doesn't, and of course this is considered a feature request rather than a bug, and no one seems to want to fix it. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Compilation after Source code changes in Asterisk
the 'make' command would typically recompile and re-link only the files that have changed. Not sure how well this works with asterisk, but I think that's the idea. Mojo Arpit Mehta wrote: hi, This might be the most obvious thing to you. I need to change some parts of the source code of Asterisk. I was wondering if we have to compile the whole source code again everytime using the commands (which i think might take some time to compile again) cd /usr/src/asterisk-version make make install or is there a faster and better way to do things Thanks a lot for all the help i have recievied from this mailing list. -- AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FX Dialing Odd
You want Asterisk to dial 1-630-XXX-XXX, but the telco only got 630-XXX-, which the provider will see 630- and assume your current area code of 312. Rob Schall wrote: But if this was the case, then why would the message playback (from the provider) read back the digits from the start. I mean, I dialed 630-XXX-XXX and it reads back 312 (my local area code)-630-XXX-X I would think if it wasn't waiting, then it would do like a 312-30X-XX-XX Rob John Novack wrote: Rob Schall wrote: Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However,every 4 or 5 times, we get an error back from the provider that says The number you have dialed. blah blah blah. Common defect in the Zaptel driver. It does NOT listen for dial tone, so if you have not inserted a w or three into the dial string, it will dial before the Central Office is ready, and it may miss a digit, causing misdials. Curious that cheap modems years ago learned to listen for dial tone, but the Zaptel driver doesn't, and of course this is considered a feature request rather than a bug, and no one seems to want to fix it. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING
Humberto Figuera schrieb: HI Tobias, look in www.soft-switch.org/unicall/unicall/index.html ;p Thank you. Not very complete but it has given me an idea what to think of unicall. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Debug meetme
Hi, I'm having complaints from some users about calls into dynamic meetme sessions failing. I'm guessing that they are dialling the wrong DTMF keys, OR that DTMF is hearing the digits entered wrong (or not hearing some). I've put debug = debug into logging.conf, and searched through the file, but I'm not sure how to debug. EG, Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'USER ABC 2060' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2060' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2098' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'from-sip' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'SIP/460-b7310bf0' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '(null)' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'MeetMe' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '|DsM' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2007-06-01 14:31:51' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2007-06-01 14:31:51' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '2007-06-01 14:32:33' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '42' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '42' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'ANSWERED' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is 'DOCUMENTATION' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '(null)' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '1180704711.1969' Jun 1 14:32:33 DEBUG[14820] pbx.c: Function result is '(null)' Where would it show what DTMF they entered? Cheers, Adrian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] no ringing tone making attended transfer whith an IAX client
Hi I have configured attended transfer in features.conf like this [general] parkext = 70 ; What ext. to dial to park parkpos = 00-99; What extensions to park calls on context = parkedcalls ; Which context parked calls are in parkingtime = 300 ; Number of seconds a call can be parked for (default is 45 seconds) featuredigittimeout = 1000 courtesytone = beep xfersound = beep ; to indicate an attended transfer is complete xferfailsound = beeperr [featuremap] blindxfer = #9 automon = *1 ; One Touch Record atxfer = # When I'm making a transfer, just after dialing the transfering number, I don't listen any ringing tone, but the transfer is made correctly. In detail, when I press # I hear Transfer and the dial tone is played. I dial the extension I want to transfer and I don't hear any ringing tone but the transfer is beeing made. Is this OK? Thank you very much. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Delay in posting of messages to list
Can anyone enlighten me as to why it takes 40 minutes or more for a posting to the list to appear. This seems excessive, as other forums do not take this long. Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FX Dialing Odd
OPTIONS to the Dial line (at the end) are different the special digits in the number string. Rob Schall wrote: Also, in the dial command the w says its for the *1 recording. Not waiting. Is the documentation wrong? What is the correct way to wait in the dial command? Rob John Novack wrote: Rob Schall wrote: Here's a possible bug, or more likely, I'm just missing something. We have a pots card in one of our asterisk boxes. Its a simple asterisk setup with one FXO/FXS card and basic static extensions file, etc. When we dial out over the pots line, 4 out of 5 times, it will work. However,every 4 or 5 times, we get an error back from the provider that says The number you have dialed. blah blah blah. Common defect in the Zaptel driver. It does NOT listen for dial tone, so if you have not inserted a w or three into the dial string, it will dial before the Central Office is ready, and it may miss a digit, causing misdials. Curious that cheap modems years ago learned to listen for dial tone, but the Zaptel driver doesn't, and of course this is considered a feature request rather than a bug, and no one seems to want to fix it. John Novack ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FX Dialing Odd
John Novack wrote: Not option w add a w in the DIAL STRING Guess you didn't search the archives? Sometime in 1.2 this feature was fixed to work with pulse dial as well. If he had searched the archives, he never would have posted the message in the first place. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Centos kernel source
I am using centos 4.4 server cd ,when I am trying to compile zaptel 1.4.2 ... error appears From where I can get the missing rpms .or kernel source From where I can get the centos 4.4 server kernel source. Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calls being dropped
We have the latest version of asterisk, on a xeon dell server (2gb ram), with 6 snom320's(latest firmware) and 3 grandstream gxp 2000's (latest stable firmware) and are having a few problems. We have a basic menu that transfers calls to different extensions. The problems can be found on all extensions. We have 2 different incoming providers and the problem happens on both providers. I want your input on 2 problems, they are the following: 1. 60% of the time everything works fine and there are no problems, 40% of times when the calls are transferred to an extension, after a few seconds , the call drops. The log from the server is below(this is from pickup to hangup, the main area of concern is where it says warning). -- Executing [EMAIL PROTECTED]:1] Answer(SIP/9097406868-09e110f8, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(SIP/9097406868-09e110f8, menus/welcome-to-exec) in new stack -- SIP/9097406868-09e110f8 Playing 'menus/welcome-to-exec' (language 'en') == CDR updated on SIP/9097406868-09e110f8 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/9097406868-09e110f8, SIP/103|50|m) in new stack -- Called 103 -- Started music on hold, class 'default', on SIP/9097406868-09e110f8 -- SIP/103-09dedd68 is ringing [May 29 09:05:22] WARNING[2678]: chan_sip.c:1899 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 1 (Critical Response) [May 29 09:05:22] WARNING[2678]: chan_sip.c:1916 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. -- Stopped music on hold on SIP/9097406868-09e110f8 == Spawn extension (from-sip, 103, 1) exited non-zero on 'SIP/9097406868-09e110f8' 2. When a call comes in or is transferred(not on outgoing), there is a delay until the person on the incoming line can hear you. We can hear them, but they can't hear us. Sometimes there is no delay, sometimes for person calling in cant hear you for 6 seconds. Thanks for the help in advance!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] cisco 7940 and auto-answer (aastra 480i vs 7940)
Having scoured the web, I still am no better off .. I have 2 Aastra480i's , and 120+ cisco 7940's :) . I am trying to decide which model to use going forward when we purchase more kit. They both seem much on a par regarding features. Q1: Is there anyway of making the cisco auto-answer _without_ having to manually edit the configuration on each phone ? I've been able to get the 480i to auto-answer by putting SIPAddHeader(Alert-Info: \;info=alert-autoanswer) into the dialplan. But it doesn't work with the cisco phones. Q2: Is there any real difference between the 480i and 9112 / 9113 phones apart from number of lines and display size ? Julian. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] realtime ldap peer matching
Hi everyone, in ldap realtime sip peers i need fullcontact set to sip:[EMAIL PROTECTED] for asterisk to correctly match the peers (at least for the natted peers to reach them)... anyway, how do I populate fullcontact on the fly with information from exten and userip? of course, i could just do it staticaly on ldap but since the info is already there why not make use of it? on res_ldap.conf i have attribute = fullcontact = AstAccountFullContact it would be nice to have something like: attribute = fullcontact = sip:.$AstExten.@.$AstIPaddress or some kind of dialplan scripting to archieve this... cheers, Caio. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk 1.4.4 Segfaults with asterisk-ooh323 from addons-1.4.1
the config file is basically the sample file gdb of the core files show the below. Loaded symbols for /usr/lib/asterisk/modules/chan_ooh323.so #0 0xb7d203e7 in strcasecmp () from /lib/libc.so.6 Any suggestions? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay in posting of messages to list
Well, the claim of 1 members MIGHT have something to do with it! E-mail delivery is notoriously erratic as well. If you think this is slow, try some on Yahoogroups! John Novack David Boyd wrote: Can anyone enlighten me as to why it takes 40 minutes or more for a posting to the list to appear. This seems excessive, as other forums do not take this long. Dave ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Delay in posting of messages to list
On 6/4/07, David Boyd [EMAIL PROTECTED] wrote: Can anyone enlighten me as to why it takes 40 minutes or more for a posting to the list to appear. This seems excessive, as other forums do not take this long. Dave - my postings consistently show up nearly immediately. Perhaps your SMTP server(s) is causing some delay. -erik ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] answer a voip call, play info.
Hi all - Not really sure where to post this question as I am just starting to research this issue. We want to allow users to dial into our did voip number. Our service will: 1. get their phone number via caller ID. look up data with the caller id. 2. generate a wave file based on the data returned play it to the user over the established voip link. How might this be done using DID origination and a VOIP stack? Or is this even what I want? Any ideas? Thank you-- Matt Pease ParkingHero, Inc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] cisco 7940 and auto-answer (aastra 480i vs 7940)
On 6/4/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: Q2: Is there any real difference between the 480i and 9112 / 9113 phones apart from number of lines and display size ? I have no experience with the Cisco's, so I can't answer those questions. However, I have deployed quite a few Aastras... I have a 480i CT on my desk which I love. Specifically, I prefer the angle that it sits up at on my desk when compared with the 9133i. I went with 9133i's throughout the rest of the office, mostly due to cost. The 480i an the 9133i both have native Power over Ethernet - no external dongles and such are needed - just patch the phone into your PoE switch and you're ready to go. The 9112i is a much more basic phone - no built-in PoE, less line appearances, and the screen doesn't have a tilt adjustment like the 9133i does. All these Aastra phones do share a common config file format though, which makes provisioning over TFTP... pretty trivial (pardon the pun). -- Justin Moore aka wantmoore ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Detecting card on the PCI Slot
I have some Analog card on a PCI slot of a remote computer, Is their a way I can figure out remotely the name of the card. I have FC6 installed on the machine. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Help with IAX
I finally got a chance to investigate this further. The fundamental
problem seemed to be that I was using a context name of iax-trunk.
When I changed this to intrunk, it worked. What are the rules for
context names? Was it the length or the special character that caused me
problems?
Thanks for your help.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David
Ruggles
Sent: Wednesday, May 30, 2007 3:10 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Help with IAX
Well this may not feel like progress, but it is. You no longer have an
authentication issue, you now have a routing issue. Could you attach a
copy of the extension.conf file on 192.168.253.21?
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network Engineer Safe Data, Inc.
(910) 285-7200[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Malcom
Kemp
Sent: Wednesday, May 30, 2007 3:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [asterisk-users] Help with IAX
From 192.168.253.20:
*CLI [May 30 14:39:03] NOTICE[6051]: chan_iax2.c:7146
socket_process: Rejected connect attempt from 192.168.253.21, request
'[EMAIL PROTECTED]' does not exist
From 192.168.253.21:
[May 30 14:39:03] WARNING[28640]: chan_iax2.c:6959
socket_process: Call rejected by 192.168.253.20: No such
context/extension
I even changed the extension to take the pattern off:
exten = 205,1,Macro(voicemail,${E205})
+
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[asterisk-users] Get calling channel before pickup
Hi, Is it possible to get the remote channelname that will be bridged when the call is answered, only having the channel that is in the Ring(ing) state? As far as I can see no variable seems to fit when doing the show channel command. I want to be able to redirect/manipulate an incoming call before it gets answered/bridged, but to do that I have to now which channel to use. Is there a way? Prefarably using manager API. Thanks, Marcus ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting card on the PCI Slot
modprobe the default analog drivers. then dmesg. -bk - Original Message - From: Sanjay Rajdev [EMAIL PROTECTED] To: asterisk-users asterisk-users@lists.digium.com Sent: Monday, June 4, 2007 12:29:37 PM (GMT-0800) America/Tijuana Subject: [asterisk-users] Detecting card on the PCI Slot I have some Analog card on a PCI slot of a remote computer, Is their a way I can figure out remotely the name of the card. I have FC6 installed on the machine. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting card on the PCI Slot
On Tue, 5 Jun 2007, Sanjay Rajdev wrote: I have some Analog card on a PCI slot of a remote computer, Is their a way I can figure out remotely the name of the card. I have FC6 installed on the machine. Try the 'lspci' command. Eg: :00:14.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface is what a Digium TDM400P card looks like. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting card on the PCI Slot
On Tue, Jun 05, 2007 at 12:59:37AM +0530, Sanjay Rajdev wrote: I have some Analog card on a PCI slot of a remote computer, Is their a way I can figure out remotely the name of the card. I have FC6 installed on the machine. lspci What driver handles it? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Get calling channel before pickup
Marcus Carlson wrote: Is it possible to get the remote channelname that will be bridged when the call is answered, only having the channel that is in the Ring(ing) state? As far as I can see no variable seems to fit when doing the show channel command. I want to be able to redirect/manipulate an incoming call before it gets answered/bridged, but to do that I have to now which channel to use. Is there a way? Prefarably using manager API. Asterisk does not know which of the several devices you are calling will pick up. Dial(SIP/1234Zap/7IAX2/fred/555) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] cisco 7940 and auto-answer (aastra 480i vs 7940)
The 9112 also doesn't have a ethernet bridge in it. The cost of adding a switch to a local office puts this unit nearly priced as a 9133 which I deploy mostly. As for the 480...I had that..upgraded to the 57i (CT now - the wireless handset supporting one). Love it. Button feel a little different but the big screen and programmable toys make it worth the differencenot an average user set though. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Justin Moore Sent: Monday, June 04, 2007 3:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] cisco 7940 and auto-answer (aastra 480i vs 7940) On 6/4/07, Julian Lyndon-Smith [EMAIL PROTECTED] wrote: Q2: Is there any real difference between the 480i and 9112 / 9113 phones apart from number of lines and display size ? I have no experience with the Cisco's, so I can't answer those questions. However, I have deployed quite a few Aastras... I have a 480i CT on my desk which I love. Specifically, I prefer the angle that it sits up at on my desk when compared with the 9133i. I went with 9133i's throughout the rest of the office, mostly due to cost. The 480i an the 9133i both have native Power over Ethernet - no external dongles and such are needed - just patch the phone into your PoE switch and you're ready to go. The 9112i is a much more basic phone - no built-in PoE, less line appearances, and the screen doesn't have a tilt adjustment like the 9133i does. All these Aastra phones do share a common config file format though, which makes provisioning over TFTP... pretty trivial (pardon the pun). -- Justin Moore aka wantmoore ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] yum om centos
There is a yum command to install from cache and not look online. Details are in man yum.Never tried what you are doing, so i can't say for sure it would work. But you probably have to pass yum the makecache command on the first machine and install from cache on the second. But this is just a guess. yum localinstall may help as well. Hope this helps. On 6/4/07, Khaled Chehab [EMAIL PROTECTED] wrote: A lot of dependencies required for each module, I don't know the sequence of the rpms.Any way to know that. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jaswinder Singh Sent: Monday, June 04, 2007 4:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] yum om centos independently install each rpm via rpm command :-/ On 04/06/07, Khaled Chehab [EMAIL PROTECTED] wrote: I have 2 servers, one connected to internet and the other is on a private lan have no access to internet. On the first server I update the kernel by yum update And installed asterisk prerequisite module yum install gcc gcc-c++ compat-gcc-32 compat-gcc-32-c++ autoconf \ libtool make automake automake14 automake15 automake16 automake17 \ bison byacc flex libtermcap libtermcap-devel newt newt-devel \ ncurses ncurses-devel openssl-devel zlib zlib-devel krb5-devel I zipped /var/cache/yum from the first server and extract it on the second server at the same directory. On the second server I tried to update using yum update but the yum update failed. How can I do that with out connecting the second server to internet . Khaled Regards * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users * No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. Any views expressed by an individual in this electronic message do not necessarily reflect views of Xplorium or its subsidiaries and associates. This electronic message and its attachments are solely addressed to the addressee(s), and contain confidential information protected from disclosure belonging to Xplorium. If you are not the intended addressee of this electronic message and its attachments, kindly delete it immediately from your system and notify the sender by electronic mail. You must not copy this message or attachment or disclose its content to any other person. Xplorium does not guarantee the integrity of this electronic message and any of its attachments, or that they are free from computer viruses or other defects. * ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FX Dialing Odd
Thanks Eric.
That is exactly what the problem was. I actually added a D(1) instead of
the w, but either would work I'm sure. Its odd that you couldn't get a
better error than that. I mean, asterisk should be able to tell what was
actually sent and received by the other end. I shouldn't say it
should, but it would be nice if it could tell. :)
Rob
Eric ManxPower Wieling wrote:
Asterisk does NOT wait for dialtone when going offhook on an FXO port.
Asterisk is sending digits to the telco too quickly.
Add a w (.5 second wait) before the extension to be dialed. Example:
exten = _91NXXNXX,n,Dial(${TRUNK}/w${EXTEN:${TRUNKMSD}},,wW)
Rob Schall wrote:
Here's a possible bug, or more likely, I'm just missing something.
We have a pots card in one of our asterisk boxes. Its a simple asterisk
setup with one FXO/FXS card and basic static extensions file, etc. When
we dial out over the pots line, 4 out of 5 times, it will work. However,
every 4 or 5 times, we get an error back from the provider that says
The number you have dialed. blah blah blah. The interesting there
there, is that the number it quotes us, adds our local area code in
front of it. And this only seems to happen on long distance context
numbers. So when we dial 630-XXX-, sometimes we are sent to
312-681-XXX-.
Here's our configs:
zaptel.conf
loadzone=us
defaultzone=us
fxoks=1-2
fxsks=3
zapata.conf
[trunkgroups]
[channels]
language=en
context=internal
usecallerid=yes
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
callgroup=1
pickupgroup=1
immediate=no
;define channels
context=internal
signalling=fxo_ks
channel = 1-2
context=zapchans
signalling=fxs_ks
group=1
channel = 3
extensions.conf
[general]
static=yes
writeprotect=no
clearglobalvars=no priorityjumping=no
[globals]
CONSOLE=Console/dsp ; Console interface
for demo
;TRUNK=Zap/g1 ; Trunk interface
TRUNK=Zap/3 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
[trunkint]
exten = _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten = _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
exten = _91NXXNXX,1,Verbose(LD)
exten = _91NXXNXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,wW)
exten = _91NXXNXX,n,Hangup()
[trunklocal]
exten = _9NXX,1,Verbose(LOCAL)
exten = _9NXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}},,wW)
exten = _9NXX,n,Hangup()
[trunktollfree]
exten = _91800NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91888NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91877NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten = _91866NXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[outgoing]
include = trunklocal
include = trunktollfree
include = trunkld
;ZAP Channels
[zapchans]
exten = 3,1,Dial(ZAP/1-1)
exten = 3,2,Hangup()
exten = 4,1,Dial(ZAP/2-1)
exten = 4,2,Hangup()
exten = s,1,Answer()
exten = s,2,NoOp(${CHANNEL:4:1})
exten = s,3,Goto(${CHANNEL:4:1},1)
exten = s,4,Hangup()
[incoming]
include =internal
[internal]
include = outgoing
exten = ,1,Dial(Zap/3/1630XXX,,wW)
exten = ,2,Hangup()
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Re: [asterisk-users] Detecting card on the PCI Slot
I have installed FC6 on it, want to configure it with Asterisk. It had some driver earlier but the machine has been formatted yesterday, so no idea. Also I am new to Linux. Regards, Sanjay Rajdev Phone : +1 (877) 342 2329 x 1702 Fax : +1 (815) 261 5907 http://www.featherstoneinformatics.com Communications from Featherstone Informatics Group (FIG) may transmit information that is confidential and privileged information of Featherstone Informatics Group (FIG). Unless you are the intended addressee, you may not use, copy or disclose to anyone this communication or any information transmitted by this communication. If you have received such communication in error, please advise the sender by e-mail and/or telephone and destroy this communication immediately. This communication and any information transmitted by this communication may also be considered protected health information as defined under the Health Insurance Portability and Accountability Act and its related regulations (a.k.a., HIPAA) or any other similar state law. Please exercise due care and ensure that you comply with its contractual and legal obligations. - Original Message - From: Tzafrir Cohen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Tuesday, June 5, 2007 2:19:22 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Detecting card on the PCI Slot On Tue, Jun 05, 2007 at 12:59:37AM +0530, Sanjay Rajdev wrote: I have some Analog card on a PCI slot of a remote computer, Is their a way I can figure out remotely the name of the card. I have FC6 installed on the machine. lspci What driver handles it? -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Detecting card on the PCI Slot
Thanks for the suggestion, I figured out the cards. I have 2 Digium TDM400P card and a Sangoma A101 single port card on the machine. Any suggestion on installing them. Regards, Sanjay Rajdev - Original Message - From: Gordon Henderson [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Tuesday, June 5, 2007 2:11:47 AM (GMT+0530) Asia/Calcutta Subject: Re: [asterisk-users] Detecting card on the PCI Slot On Tue, 5 Jun 2007, Sanjay Rajdev wrote: I have some Analog card on a PCI slot of a remote computer, Is their a way I can figure out remotely the name of the card. I have FC6 installed on the machine. Try the 'lspci' command. Eg: :00:14.0 Communication controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface is what a Digium TDM400P card looks like. Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] debug logs
This notifies you that it has been used (IIRC). Hi what does that mean , it has been IIRC ? ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] realtime ldap peer matching
On 04/06/07, Caio Zanolla [EMAIL PROTECTED] wrote: Hi everyone, in ldap realtime sip peers i need fullcontact set to sip:[EMAIL PROTECTED] for asterisk to correctly match the peers (at least for the natted peers to reach them)... anyway, how do I populate fullcontact on the fly with information from exten and userip? Wouldn't these just be dialplan vars? of course, i could just do it staticaly on ldap but since the info is already there why not make use of it? on res_ldap.conf i have attribute = fullcontact = AstAccountFullContact it would be nice to have something like: attribute = fullcontact = sip:.$AstExten.@.$AstIPaddress or some kind of dialplan scripting to archieve this... I'm pretty sure res_ldap.c can't do this yet. What version (* and res_ldap) and schema are you using btw? IIRC, the latest version doesn't need: attribute = fullcontact = AstAccountFullContact just: fullcontact = AstAccountFullContact Thanks, Gavin. cheers, Caio. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] debug logs
means (I)f (I) (R)emember (C)orrectly On 6/4/07, ram [EMAIL PROTECTED] wrote: This notifies you that it has been used (IIRC). Hi what does that mean , it has been IIRC ? ram ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Auto Dial Problem
Dear Sir,
Thank you very much
ASLAY
- Original Message -
From: Nasir Iqbal [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, June 03, 2007 5:43 PM
Subject: Re: [asterisk-users] Auto Dial Problem
Hi,
I setup auto dial on my asterisk server. The problem
is asterisk does not wait for called party to answer
the call but proceed to process the extension specifed
in my .call file
No problem with Auto Call
exten = _01N.,1,Dial(Zap/g1/${EXTEN},20)
the problem with zap channel
try callprogress with yes in zapata.conf
It may cause another problem, after remote party has picked up the
call and asterisk still does not know it. and in ringing status.
if your dial plan work fine now, then no need to change rxgain.
otherwise.
Just Increase your rxgain value. try with different values and choose
best one.
if rxgain greater then desired value ?? you my receive invalid report
that remote party has picked up.
if rxgain less then desired value ?? you my receive invalid ringing
report after call is answered.
so adjust it according your requirement and also check noise and quality
your PSTN lines.
Regards
Nasir Iqbal
ICT Innovations
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Re: [asterisk-users] Calls being dropped
that becasue the reinvite is using a private ip probably.. sip debug pastebin the results.. look in the re-invite part.. On 6/4/07, Compnet Bobby [EMAIL PROTECTED] wrote: We have the latest version of asterisk, on a xeon dell server (2gb ram), with 6 snom320's(latest firmware) and 3 grandstream gxp 2000's (latest stable firmware) and are having a few problems. We have a basic menu that transfers calls to different extensions. The problems can be found on all extensions. We have 2 different incoming providers and the problem happens on both providers. I want your input on 2 problems, they are the following: 1. 60% of the time everything works fine and there are no problems, 40% of times when the calls are transferred to an extension, after a few seconds , the call drops. The log from the server is below(this is from pickup to hangup, the main area of concern is where it says warning). -- Executing [EMAIL PROTECTED]:1] Answer(SIP/9097406868-09e110f8, ) in new stack -- Executing [EMAIL PROTECTED]:2] BackGround(SIP/9097406868-09e110f8, menus/welcome-to-exec) in new stack -- SIP/9097406868-09e110f8 Playing 'menus/welcome-to-exec' (language 'en') == CDR updated on SIP/9097406868-09e110f8 -- Executing [EMAIL PROTECTED]:1] Dial(SIP/9097406868-09e110f8, SIP/103|50|m) in new stack -- Called 103 -- Started music on hold, class 'default', on SIP/9097406868-09e110f8 -- SIP/103-09dedd68 is ringing [May 29 09:05:22] WARNING[2678]: chan_sip.c:1899 retrans_pkt: Maximum retries exceeded on transmission [EMAIL PROTECTED] for seqno 1 (Critical Response) [May 29 09:05:22] WARNING[2678]: chan_sip.c:1916 retrans_pkt: Hanging up call [EMAIL PROTECTED] - no reply to our critical packet. -- Stopped music on hold on SIP/9097406868-09e110f8 == Spawn extension (from-sip, 103, 1) exited non-zero on 'SIP/9097406868-09e110f8' 2. When a call comes in or is transferred(not on outgoing), there is a delay until the person on the incoming line can hear you. We can hear them, but they can't hear us. Sometimes there is no delay, sometimes for person calling in cant hear you for 6 seconds. Thanks for the help in advance!!! ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mike Sales Manager http://www.voicemeup.com Making it happen 1.877.807.VOIP (8647) 1.514.312.7030 ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Oddity
I have two Asterisk servers. One is my primary server that I link to all of my providers and the other is at an office building with multiple tenants. If I tell Asterisk to dial an entry in the iax.conf that is for one customer off that second box, why does it use a different account for a different customer? It still ends up at the correct box, but it is hard to troubleshoot issues when calls are coming in under different accounts (probably different account codes too). Also, the second customer on that box. Earlier today everything worked fine as was. Later all calls going to that customer were going to the default context, despite the fact that I explicitly defined the context I wanted the calls to go to in all entries in iax.conf. - Mike Hammett Intelligent Computing Solutions http://www.ics-il.com http://www.ics-il.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
