[asterisk-users] Problem in placing Call with Asterisk (Got SIP response 500 Internal Server Error)
Hi guys this is my Ist mail on this group, I am running asterisk with CentOS 4.4 machine. When i initiate a call then error message apears. calling Number is provided to Asterisk by the php application. Error message appears like this Got SIP response 500 Internal Server Error back from 209.47.92. [Oct 10 23:55:20] WARNING[19289]: pbx.c:4976 ast_pbx_outgoing_exten: SIP/verizon1-00ad5ba0 already has a call record?? Channel SIP/verizon1-00ad5ba0 was never answered plz help me in resolving this issue. waiting for ur response. -- Syed Jamshed Zaidi (Jamy-Virus) Linux Admin/Programmer @ Naseeb Networks 0321-4087492 Shoot for the moon. Even if you miss, you'll land among the stars ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinions on Release Numbering
On Wed, 10 Oct 2007 12:54:42 -0500, Russell Bryant wrote: I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc. Another proposal has been using 1.5 to indicate that it is a release candidate. For example, 1.5.3, 1.5.3.1, 1.5.3.2, etc., would be the release candidates for the upcoming 1.6.3 release. the former is more obvious than the latter. i kind of like asterisk's release numbering mechanism where the even numbered dot releases are stable/production while the odd numbered ones are for development. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinion on hardware (computer) for an Asterisk Server!
On Wed, 10 Oct 2007, Raúl Gómez C. wrote: Hi list, I'm about to install Asterisk on an Old HP NetServer LC2000 Server (year 2001), it has 2 Pentium III 1GHz CPUs (Coppermine FSB 133MHz 256K L2 Cache), 768MB PC-133 ECC RAM, 3 UltraSCSI LVD2 18.2GB 10K RPM HDD in RAID5, 100Mb NIC for server. This Server will support 35 SIP phones (users) and 10 FXO ports (for telco lines) and 2 FXS ports (internal analog phones) with a Sangoma Remora A400 PCI card. What do you think? is this hardware enough for this setup??? Yes - You should be fine. (Based on my own use of 1GHz Via boards) However, it's OLD. 6 years old now. What's going to fail first? The drives? PSU? Fans? Are you going to put your company's phone system which has to just work on an aging server? Do yourself a favour and spend some money on a modern box. You're probably going to spend somewhere in the region of £3500 on the phones themselves (probably more), so even a quarter of that will get you a good modern box to run it all on (Which with the right processor and drive configuration will probably suck less power too, if that's a concern for you) Gordon___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Understanding RTCP in Asterisk
X-ECN Telecoms-MailScanner-Information: Please contact ECN Telecoms for more information X-ECN Telecoms-MailScanner: Found to be clean X-ECN Telecoms-MailScanner-From: [EMAIL PROTECTED] X-Spam-Status: No My third try, humph! Yusuf wrote: Hi, I am trying to understand the RTCP stats in Asterisk. 1. I am using the 'h' exten to store the RTCP records in CDRS. However, only if the caller hangups does the RTCP values have anything in them. If the caller hangups, the values gets stored in CDRs, but they all empty(0). So even on the CLI, I can see that the values for RTCP get completed if the caller hangs up, but if the callee hangs up the values are all zero. 2. I have values for Jitter and packet loss, however the RTT is always 0. I am using http://bugs.digium.com/view.php?id=10590 with 1.4.11, which makes available the RTCP stats for the whole call, not only the last packet, which is the general behavior of stable asterisk 1.4.x -- thanks, Yusuf ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there a way to turn off SIP METHOD OPTIONS in asterisk ?
Hi, is there a way to turn of SIP METHOD OPTIONS in asterisk? I have a sip pbx which ignore Sip Option Messages from a unknown user. Asterisk send Option Messages to peers with From: [EMAIL PROTECTED] The sip server expects From: [EMAIL PROTECTED] server domain]. So i have to turn off Options Messages or to set a correct From: . Any ideas? THX ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Polycom IP Phones and Asterisk
Hi List; I am trying to find a link to see the polycom IP Phones that work with Asterisk, but not able to find until now. I checked this link, but did not find any thing related to Polycom IP Phones: http://www.voip-info.org/wiki/view/Asterisk+phones So any advise where I can find a link to see the IP Phones of Polycom and its configurations? Regards Bilal Moody friends. Drama queens. Your life? Nope! - their life, your story. Play Sims Stories at Yahoo! Games. http://sims.yahoo.com/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Randomly half-voice at sip/zap
Ok, so i made the terminal screen wider, but during the call nothing changes: ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ###* Rx: 10736 (10736) Tx: 0 (0) What could be the reason? THx 2007/10/10, Mojo with Horan Company, LLC [EMAIL PROTECTED]: Péter Tóth wrote: When i try ztmonitor as follows, it gives strange output... [EMAIL PROTECTED] zaptel-1.2.10]# ./ztmonitor 1 -vv Visual Audio Levels. Use zapata.conf file to adjust the gains if needed. ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ###* R ###* R If ztmonitor keeps scrolling down the screen, you need to make your terminal wider. The '#' marks should jump back and forth left and right like a level monitor, and there will only be one row of them (but with two levels, one for RX and one for TX). The screen won't scroll at all. Try this again :) Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ Tóth Péter Tel.: +36703834578 _ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom IP Phones and Asterisk
bilal ghayyad wrote: I checked this link, but did not find any thing related to Polycom IP Phones: http://www.voip-info.org/wiki/view/Asterisk+phones So any advise where I can find a link to see the IP Phones of Polycom and its configurations? Regards Bilal If you type polycom in the search box on the voip-info.org page you get to this page: http://www.voip-info.org/wiki/view/Polycom+Phones Al -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] How are you using Asterisk at Home ?
Totally agree *IF* the SIP elements behind your router/firewall have real IP addresses and you are not using NAT in your router. With NAT scenarios, I prefer to have a copy of Asterisk running on firewall/NAT router so it at least has one public IP address to make various SIP games a little easier. iptables can really protect asterisk from uninvited (npi) SIP / RTP packets if you are really paranoid also the asterisk running on your firewall/NAT router can be dedicated to just gateway functions and have your important and private asterisk pbx behind the NAT/firewall using the gateway as needed On 10/10/07, Steve Prior [EMAIL PROTECTED] wrote: Repeat after me - NEVER NEVER NEVER run other servers on your router/firewall machine!!! ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Buying Polycom
Hi List; Any one can advise me to a good link to see and buy Polycom IP Phones? Also, if I need support (in case the Phone was damaged and need to replace, so the warantee), so which web can provide that? I do not need to buy from one and he is not responsible for support. Regards Bilal Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545433 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] GTALK problem
Hi, I installed gtalk on asterisk 1.4.12.1, I change on rtp.conf the port from 1000 to 4 If I calling asterisk with GTALK in english everything is ok, however, some of my friends with the italian version of gtalk they cannot have the audio. Is it a bug? Or I did some mistake ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Congested/busy
hi all i have a TE110P connected to my PBX when i try to call a extension number in other location 3525 the asterisk give me a error -- User entered '3525' -- Executing [EMAIL PROTECTED]:4] GotoIf(Zap/31-1, 0?6:5) in new stack -- Goto (lacnicuy,450,5) -- Executing [EMAIL PROTECTED]:5] Dial(Zap/31-1, IAX2/lacnic:[EMAIL PROTECTED]/3525|30|r) in new stack -- Called lacnic:[EMAIL PROTECTED]/3525 -- IAX2/nicbr-1 is circuit-busy [Oct 11 10:08:02] NOTICE[2763]: chan_iax2.c:2925 __auto_congest: Auto-congesting call due to slow response -- Hungup 'IAX2/nicbr-1' [Oct 11 10:08:02] NOTICE[2781]: cdr.c:434 ast_cdr_free: CDR on channel 'IAX2/nicbr-1' not posted == Everyone is busy/congested at this time (1:0/1/0) -- Executing [EMAIL PROTECTED]:6] Hangup(Zap/31-1, ) in new stack == Spawn extension (lacnicuy, 450, 6) exited non-zero on 'Zap/31-1' -- Hungup 'Zap/31-1' anybody can help me with this? thanks ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Buying Polycom
www.telephonydepot.com has good prices. Never needed their support so I can't comment www.voipsupply.com a bit more expensive than above. Great support - Original Message - From: bilal ghayyad [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Thursday, October 11, 2007 12:08 PM Subject: [asterisk-users] Buying Polycom Hi List; Any one can advise me to a good link to see and buy Polycom IP Phones? Also, if I need support (in case the Phone was damaged and need to replace, so the warantee), so which web can provide that? I do not need to buy from one and he is not responsible for support. Regards Bilal Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=listsid=396545433 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] OpenVox A400P01 not detected
Hello Has someone used the OpenVox A400P01 (ie. a supposedly Digium-compatible A400P board with a single FXO module www.openvox.com.cn/products_detail.php?genre_id=9id=28) successfully? I've put it in an older PC with a Gigabyte GA-7ZX motherboard, then a more recent PC with an Asrock K8NF4G-SATA2: dmesg returns nothing :-/ Is there something specific that needs to be done in either the BIOS or Linux to get this board to work? Thank you. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinions on Release Numbering
On Wednesday 10 October 2007 12:54:42 Russell Bryant wrote: I have been having discussions with various members of the development community in regards to changes to the way we manage open source Asterisk releases. The changes that we eventually decide on will determine how we manage the 1.6 version of Asterisk. I will be posting much more detailed information about 1.6 in the near future. What I'm looking for right now is some opinions on version numbering. Part of the working plan for Asterisk 1.6 involves making release candidates for every 1.6.X release, so that various community members can help with doing regression testing on the changes before making the release. I proposed calling the release candidates 1.6.3-rc1, 1.6.3-rc2, etc. One of the problems with this traditional approach is that it's not obvious unless you know what rc means. In the case of someone new to software development, I want them never to assume that 1.6.0-rc2 means 1.6.0 plus something else, presumably desireable to have. Note that this isn't without precedence; netatalk was distributed for years as netatalk-1.3+asun. It would be perfectly reasonable to assume that rc was someone's initials. Another proposal has been using 1.5 to indicate that it is a release candidate. For example, 1.5.3, 1.5.3.1, 1.5.3.2, etc., would be the release candidates for the upcoming 1.6.3 release. This method is no less obvious than rc1 for the untrained and ensures that people who do not wish to become guinea pigs will remain out of that arena (i.e. if they only choose the version that sorts to the bottom of the directory, they will always be running a release). The universal problem is that we'd like people who know little to pick the right version, with no training (and yes, the system using rc to indicate release candidates is also a matter of training, the abbreviation is not obvious to the untrained). -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GTALK problem
If I calling asterisk with GTALK in english everything is ok, however, some of my friends with the italian version of gtalk they cannot have the audio. Audio problems might be experienced with older Gtalk clients. Version 1.0.0.104 is reported to work. The following resources may help you : http://www.voip-info.org/wiki/view/Asterisk+Google+Talk#Bugsampknownissues http://bugs.digium.com/view.php?id=10512 Hope this will help you solve the problem, Philippe ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinions on Release Numbering
On Thu, Oct 11, 2007 at 08:47:52AM -0500, Tilghman Lesher wrote: One of the problems with this traditional approach is that it's not obvious unless you know what rc means. In the case of someone new to software development, I want them never to assume that 1.6.0-rc2 means 1.6.0 plus something else, presumably desireable to have. Note that this isn't without precedence; netatalk was distributed for years as netatalk-1.3+asun. It would be perfectly reasonable to assume that rc was someone's initials. That someone could be apt/yum or rpm/deb trying to figure out the latest version of the package to upgrade to. There are some common workarounds. And they all require some manipulations to the version number as recieved from the tarball before packaging it. Anyway, following that logic, go for 1.5.99-rc2 ? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinions on Release Numbering
Tilghman Lesher wrote: This method is no less obvious than rc1 for the untrained and ensures that people who do not wish to become guinea pigs will remain out of that arena (i.e. if they only choose the version that sorts to the bottom of the directory, they will always be running a release). The universal problem is that we'd like people who know little to pick the right version, with no training (and yes, the system using rc to indicate release candidates is also a matter of training, the abbreviation is not obvious to the untrained). Can I chip in my comments here? There are some defacto standards for release numbering. rcX for pre-releases and pure numerical for releases is one (probably the most widely used) Odd/Even numbering for stable/unstable. Personally, I nave no overriding preference, but the rcX nomenclature is far more obvious than the odd/even scenario. Secondly, would any of the people who know little really be downloading software (probably in source form) without having read about it first? And, the status of any release of software is almost always documented and publicised when it appears anyway... Either on the front page for the download area or via google ;-) Hope you don't mind me chipping in... I'm really enjoying getting to grips with Asterisk. It's great! Cheers Alan -- The way out is open! http://www.theopensourcerer.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Difference between trunk and released versions
Hello, Up to a while ago I thought that the released versions are checkpoints of the trunk versions; however, now I understand they are not, as I see differences between the two trains. So, what is the relation between them? Examples for differences: - When the language is different than Engligh the trunk version is reading numbers from /var/lib/asterisk/sounds/Lang-Name/digits while the release version is using /var/lib/asterisk/sounds/digits/Lang-Name - MAILBOX_EXISTS function is replaced with MailboxExists application. - External IVR has no way to exit from the program under the release version... The documentation is correct with the trunk version. Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinion on hardware (computer) for an Asterisk Server!
Good point Gordon, but I have 2 spare drives (of line), the server has 2 (redundant) PSU, one of this brand new, the fans has already failed and has bee replaced, so there are brand new too. I'm not sure if a server has another component that is prone to fail, so any advise/suggestion is welcome. Thanks Gordon! On 10/11/07, Gordon Henderson [EMAIL PROTECTED] wrote: Yes - You should be fine. (Based on my own use of 1GHz Via boards) However, it's OLD. 6 years old now. What's going to fail first? The drives? PSU? Fans? Are you going to put your company's phone system which has to just work on an aging server? Do yourself a favour and spend some money on a modern box. You're probably going to spend somewhere in the region of £3500 on the phones themselves (probably more), so even a quarter of that will get you a good modern box to run it all on (Which with the right processor and drive configuration will probably suck less power too, if that's a concern for you) Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Exposing sound files through http for links
Hi. I'd like for my sound files to be exposed through http. You know, the ones located in var/lib/asterisk/sounds. This is probably an apache thing i'd have to configure or is accessible through some asterisk http routing? 1. how one would configure this? 2. what are the security costs of doing this to asterisk? - Dominic ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Buying Polycom
On Thu, 2007-10-11 at 03:08 -0700, bilal ghayyad wrote: Hi List; Any one can advise me to a good link to see and buy Polycom IP Phones? Also, if I need support (in case the Phone was damaged and need to replace, so the warantee), so which web can provide that? I do not need to buy from one and he is not responsible for support. You did not say were you are located so here's a suggestion for a US company that sells Polycom via the web and does warranty/support: http://www.voipsupply.com Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exposing sound files through http for links
On 10/11/07, Dominic Son wrote: Hi. I'd like for my sound files to be exposed through http. You know, the ones located in var/lib/asterisk/sounds. This is probably an apache thing i'd have to configure or is accessible through some asterisk http routing? 1. how one would configure this? 2. what are the security costs of doing this to asterisk? assuming you are using apache to do this, change the ServerRoot directive in your httpd.conf which can in /etc/httpd/conf/ for a Centos install. alternately you can just create a link, eg if apache root is/home/son/www create a link to sounds ln -s /var/lib/asterisk/sounds /home/son/www/sounds then pointing your browser to http://localhost/sounds should list everything in the sounds directory. I don't know how much of a security risk this is. If you don't need to expose the entire directory for listing but only files on a selective basis then throw in an index.html into the sounds directory with hello world in it, now files can still be accessed by someone with a link to a specific file but no fishin around. -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] OpenVox A400P01 not detected
On Thu, 2007-10-11 at 15:07 +0200, Vincent wrote: Hello Has someone used the OpenVox A400P01 (ie. a supposedly Digium-compatible A400P board with a single FXO module www.openvox.com.cn/products_detail.php?genre_id=9id=28) successfully? I've put it in an older PC with a Gigabyte GA-7ZX motherboard, then a more recent PC with an Asrock K8NF4G-SATA2: dmesg returns nothing :-/ Is there something specific that needs to be done in either the BIOS or Linux to get this board to work? That board is 100% compatible with the Digium TDM400P so you only need to load the wctdm module to activate it and configure zaptel.conf and zapata.conf as usual. You should do something like this with lspci: 00:08.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Weatherproof Hard Phone
Philipp Kempgen wrote: Don Kelly wrote: http://www.sandman.com/autodial.html These phones look like the ones we had in Germany 20 years ago. ;-P Hey, don't knock it, Phillipp :) -- I'm as big a fan of German technology as anybody, but these phones are amazing pieces of engineering. Reliable, with excellent sound quality, and practically indestructible. There's a reason they're still in production after all these years. -Stephen- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] TDM400P
Hello all, i've configured a TDM400P card but some calls hangs up and when i take the phone to do a call y hear someone that callme. How is the way to check the line before to do a call?. Other thing, is there a way to use Dial application without ring the phone if the line is busy or unavailable?. I want to get busy channel or unavailable channel whitout ring or music on hold if the phone are busy or no available. Thanks!! Alejandro González ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Calls dropping...
I have a customer that recently started having a problem with their Call Center SIP extensions. The problem is that after some time the caller will hear a triple tone (beep, beep, beep), a 5 second pause, another triple tone and then the call will be dropped. This usually happens between the 8 an 10 minute mark. Until Tuesday we were running Asterisk 1.4.11 but I decided to upgrade to 1.4.12.1 just in case this was a bug with earlier versions. The problem only started recently, about a week ago. We have not made any significant changes to the configuration, mostly just dialplan changes so we do not know what exactly is causing this. Any ideas? -- Telecomunicaciones Abiertas de México S.A. de C.V. Carlos Chávez Prats Director de Tecnología +52-55-91169161 ext 2001 signature.asc Description: This is a digitally signed message part ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] $70 USD bounty for simple Junghanns ISDNguard shell script
Nick Richardson wrote: Hi all, I recently purchased a Junghanns ISDNguard and to my horror I found out: - Junghanns technical support is non-existant - I can't use it without recompiling Asterisk with res_watchdog Let me know if you get any response on this bounty. Cheers, Stephen Bosch Calgary, Alberta, Canada ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinion on hardware (computer) for an Asterisk Server!
On Thu, 11 Oct 2007, Raúl Gómez C. wrote: Good point Gordon, but I have 2 spare drives (of line), the server has 2 (redundant) PSU, one of this brand new, the fans has already failed and has bee replaced, so there are brand new too. I'm not sure if a server has another component that is prone to fail, so any advise/suggestion is welcome. Hard to say, really. Electrolytic capacitors are what tends to fail over the longer term, and I'd put in a new CMOS/RTC battery for good measure too. And I'm sure you're not alone in trying to recycle old hardware - I myself have several old systems acting as various test servers/firewalls, etc. but none that are in what I'd consider to be a mission critical position. I'd suggest it would be a good system to develop the software on, test your dial-plans, etc. but I'd still want to look at modern hardware for a live production system. Gordon Thanks Gordon! On 10/11/07, Gordon Henderson [EMAIL PROTECTED] wrote: Yes - You should be fine. (Based on my own use of 1GHz Via boards) However, it's OLD. 6 years old now. What's going to fail first? The drives? PSU? Fans? Are you going to put your company's phone system which has to just work on an aging server? Do yourself a favour and spend some money on a modern box. You're probably going to spend somewhere in the region of £3500 on the phones themselves (probably more), so even a quarter of that will get you a good modern box to run it all on (Which with the right processor and drive configuration will probably suck less power too, if that's a concern for you) Gordon ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Exposing sound files through http for links
If you are worried about it affecting asterisk, you could copy them to another web server. -- -- Steven http://www.glimasoutheast.org Dominic Son [EMAIL PROTECTED] wrote in message news:[EMAIL PROTECTED] Hi. I'd like for my sound files to be exposed through http. You know, the ones located in var/lib/asterisk/sounds. This is probably an apache thing i'd have to configure or is accessible through some asterisk http routing? 1. how one would configure this? 2. what are the security costs of doing this to asterisk? - Dominic ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Maximum manager connections
Have anyone maided like 200 simultaneous connections to asterisk AMI (manager). ?? How many connections can I made without problems ? I’m using a Quad core DELL poweredge machine. Roberto Fernandes Lopes Diretor Presidente Dialtech Telecom. e Sistemas Ltda. (11) 6986-8886 No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.14.8/1063 - Release Date: 11/10/2007 09:11 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Buying Polycom
I'm in Venezuela, and I have buyed over 5K$ to htt://www.voipsupply.com, excellent service and they sell warranty extensions for any product! On 10/11/07, Patrick [EMAIL PROTECTED] wrote: You did not say were you are located so here's a suggestion for a US company that sells Polycom via the web and does warranty/support: http://www.voipsupply.com Regards, Patrick ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Maximum manager connections
Use the astmanproxy and move the load elsewhere. (If you just want to passively listen to messages, your box is about 100 times faster than you need :) Zoa Roberto wrote: Have anyone maided like 200 simultaneous connections to asterisk AMI (manager). ?? How many connections can I made without problems ? I’m using a Quad core DELL poweredge machine. *Roberto Fernandes Lopes* *Diretor Presidente*** *Dialtech Telecom. e Sistemas Ltda.*** *(11) 6986-8886*** No virus found in this outgoing message. Checked by AVG Free Edition. Version: 7.5.488 / Virus Database: 269.14.8/1063 - Release Date: 11/10/2007 09:11 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Calls dropping...
Carlos Chavez wrote: I have a customer that recently started having a problem with their Call Center SIP extensions. The problem is that after some time the caller will hear a triple tone (beep, beep, beep), a 5 second pause, another triple tone and then the call will be dropped. This usually happens between the 8 an 10 minute mark. Until Tuesday we were running Asterisk 1.4.11 but I decided to upgrade to 1.4.12.1 just in case this was a bug with earlier versions. The problem only started recently, about a week ago. We have not made any significant changes to the configuration, mostly just dialplan changes so we do not know what exactly is causing this. Any ideas? Just curious, is this a calling card or prepaid kind of call center? Your configs and logs might help out alot. Thanks, Steve ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there a way to turn off SIP METHOD OPTIONS in asterisk ?
Andreas Bayer wrote: is there a way to turn of SIP METHOD OPTIONS in asterisk? I have a sip pbx which ignore Sip Option Messages from a unknown user. Asterisk send Option Messages to peers with From: [EMAIL PROTECTED] The sip server expects From: [EMAIL PROTECTED] server domain]. So i have to turn off Options Messages or to set a correct From: . Make sure you have qualify=no for the sip.conf entry for that PBX. Also make sure you do not have a mailbox= entry for that sip.conf entry as well. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Alert_INFO x2 = 400 Bad Request
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Good evening,
I have something strange, when I add an ALERT_INFO variable to a ring group,
the invite generated contains 2 lines with Alert-Info and my phones return a
400 Bad Request...
I've checked in my config files, there is only one line with
Set(__ALERT_INFO.
Any idea??
PS: I'm using Asterisk Asterisk 1.4.13-BRIstuffed-0.4.0-test4
Thanks
Alert-Info: Ringer-2.
Alert-Info: Ringer-2.
exten = 6800,1,Macro(user-callerid,)
exten = 6800,n,GotoIf($[foo${BLKVM_OVERRIDE} = foo]?skipdb)
exten = 6800,n,GotoIf($[${DB(${BLKVM_OVERRIDE})} = TRUE]?skipov)
exten = 6800,n(skipdb),Set(__NODEST=)
exten = 6800,n,Set(__BLKVM_OVERRIDE=BLKVM/${EXTEN}/${CHANNEL})
exten = 6800,n,Set(__BLKVM_BASE=${EXTEN})
exten = 6800,n,Set(DB(${BLKVM_OVERRIDE})=TRUE)
exten = 6800,n(skipov),Set(RRNODEST=${NODEST})
exten = 6800,n(skipvmblk),Set(__NODEST=${EXTEN})
exten = 6800,n,Set(__ALERT_INFO=Ringer-1)
exten = 6800,n,Set(RecordMethod=Group)
exten = 6800,n,Macro(record-enable,6740,${RecordMethod})
exten = 6800,n,Set(RingGroupMethod=ringall)
exten = 6800,n(DIALGRP),Macro(dial,7,${DIAL_OPTIONS},6740)
exten = 6800,n,Set(RingGroupMethod=)
exten = 6800,n,GotoIf($[foo${RRNODEST} != foo]?nodest)
exten = 6800,n,Set(__NODEST=)
exten = 6800,n,dbDel(${BLKVM_OVERRIDE})
exten = 6800,n,Goto(ext-group,6799,1)
exten = 6800,n(nodest),Noop(SKIPPING DEST, CALL CAME FROM Q/RG: ${RRNODEST})
#
U 192.168.95.235:5060 - 192.168.95.73:5060
INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 192.168.95.235:5060;branch=z9hG4bK7f1bf341;rport.
From: 0614730696 sip:[EMAIL PROTECTED];tag=as6aaa622f.
To: sip:[EMAIL PROTECTED]:5060;user=phone.
Contact: sip:[EMAIL PROTECTED].
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE.
User-Agent: Asterisk PBX.
Max-Forwards: 70.
Date: Thu, 11 Oct 2007 17:00:58 GMT.
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY.
Supported: replaces.
Alert-Info: Ringer-2.
Alert-Info: Ringer-2.
Content-Type: application/sdp.
Content-Length: 266.
___
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Re: [asterisk-users] Randomly half-voice at sip/zap
Péter Tóth wrote: Ok, so i made the terminal screen wider, but during the call nothing changes: ( # = Audio Level * = Max Audio Hit ) (RX) (TX) ###* Rx: 10736 (10736) Tx: 0 (0) What could be the reason? Maybe you're monitoring the wrong zap channel? If you put '1' to refer to group 1, that's not the way ztmonitor works -- it wants a specific zap channel number. Just an idea, I'm not sure why else it wouldn't work. Maybe set up a call and switch ztmonitor from channel to channel until you find the one that's in use. Further, I don't recall your setup -- make sure you're actually using a ZAP channel in your bridged call ;) Sorry if all this seems obvious, but I can't imagine any other reasons it wouldn't work. Moj ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Buying Polycom
bilal ghayyad wrote: Hi List; Any one can advise me to a good link to see and buy Polycom IP Phones? I've been using http://tritechcoa.com/ and they are always very prompt about email support, I've never had to send anything back to them though. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Is there real benefits on a SMP machine for Asterisk?
Hi list, I'm now considering to buy a new server for an Asterisk installation, since I've been kindly advisedhttp://lists.digium.com/pipermail/asterisk-users/2007-October/198146.htmlnot to use an old server for a mission critical app... Well, playing around in Dell's, HP's and IBM's online stores, I've noticed a lot of Discounts or even FREE upgrades from Dual to Quad Core CPU's, and I cant believe that Dell offers FREE upgrades in some cases from One to Two CPU's (at least in the PowerEdge 2950 and 2970 models). I've chosen the 29xx line of servers for the redundant power supply, since it is really important to keep the uptime of the PBX as high as possible! At this point I was wondering if Asterisk gets real benefits on systems with several cores (up to 8 in Dell PE2950) for a system that will handle up to 35 simultaneous SIP call with 10 FXO ports and 2 FXS for analog phones/fax (Sangoma A400D PCI card). Thanks! Raul ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinions on Release Numbering
On Thu, Oct 11, 2007 at 08:47:52AM -0500, Tilghman Lesher wrote: One of the problems with this traditional approach is that it's not obvious unless you know what rc means. In the case of someone new to software development, I want them never to assume that 1.6.0-rc2 means 1.6.0 plus something else, presumably desireable to have. Note that this isn't without precedence; netatalk was distributed for years as netatalk-1.3+asun. It would be perfectly reasonable to assume that rc was someone's initials. I disagree. For the audience has to make a choice of which version to release without asking someone more knowledgeable, it's perfectly serviceable. Anyone not smart enough to know that rc in a version number means Release Candidate shouldn't be picking their own version anyway, they should be using a package, or asking someone (which comes to the same thing). Another proposal has been using 1.5 to indicate that it is a release candidate. For example, 1.5.3, 1.5.3.1, 1.5.3.2, etc., would be the release candidates for the upcoming 1.6.3 release. This method is no less obvious than rc1 for the untrained and ensures that people who do not wish to become guinea pigs will remain out of that arena (i.e. if they only choose the version that sorts to the bottom of the directory, they will always be running a release). No, this is *much* less obvious than rc1. The universal problem is that we'd like people who know little to pick the right version, with no training (and yes, the system using rc to indicate release candidates is also a matter of training, the abbreviation is not obvious to the untrained). Certainly. People who know little should not be *trying* to interpret version numbers; they should be using what a packager, a website, or a knowledgeable other source *tells* them to use. Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinions on Release Numbering
On Thu, Oct 11, 2007 at 04:21:09PM +0200, Tzafrir Cohen wrote: Anyway, following that logic, go for 1.5.99-rc2 ? Please don't. That parses as the second release candidate for 1.5.99. Really. To everyone. I'm not much for .99 in the first place, but you get one or the other; not both. /hobbyhorse Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
At this point I was wondering if Asterisk gets real benefits on systems with several cores (up to 8 in Dell PE2950) for a system that will handle up to 35 simultaneous SIP call with 10 FXO ports and 2 FXS for analog phones/fax (Sangoma A400D PCI card). I suppose that yes. Asterisk uses pthread, and it should distribute load across multiple cores. However, i doubt that you will need that much for 35 simultenous calls. I have 8-core system that has web interface + sql + java + some other stuff running, and at 30 simultenous calls i get loadavg maximum of 3. Regards, Atis -- Atis Lezdins VoIP Developer, IQ Labs Inc. [EMAIL PROTECTED] Skype: atis.lezdins Cell Phone: +371 28806004 Work phone: +1 800 7502835 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between trunk and released versions
Yehavi, The release branches (1.2, 1.4) were at one time trunk. When it was decided to release 1.4, for example, it was branched off from trunk as the 1.4branch. New functionality continued to be added to trunk after that. Once the release branches are created, they are feature-frozen and only bug fixes are applied (this is the goal, though sometimes new functionality does sneak in when deemed necessary or desirable by the maintainers). So long story short, up until the release of 1.4, it was the same code as trunk. Since the 1.4 branch was created, no new functionality has been added to that branch, but 1.4.x releases continue to be made as bugs are discovered and fixed. Does this clear things up? Sean On Thu, 11 Oct 2007 16:34 +0200, Yehavi Bourvine +972-8-9489444 [EMAIL PROTECTED] wrote: Hello, Up to a while ago I thought that the released versions are checkpoints of the trunk versions; however, now I understand they are not, as I see differences between the two trains. So, what is the relation between them? Examples for differences: - When the language is different than Engligh the trunk version is reading numbers from /var/lib/asterisk/sounds/Lang-Name/digits while the release version is using /var/lib/asterisk/sounds/digits/Lang-Name - MAILBOX_EXISTS function is replaced with MailboxExists application. - External IVR has no way to exit from the program under the release version... The documentation is correct with the trunk version. Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Any fool can write code that a computer can understand. Good programmers write code that humans can understand. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
On 10/11/07, Raúl Gómez C. [EMAIL PROTECTED] wrote: At this point I was wondering if Asterisk gets real benefits on systems with several cores (up to 8 in Dell PE2950) for a system that will handle up to 35 simultaneous SIP call with 10 FXO ports and 2 FXS for analog phones/fax (Sangoma A400D PCI card). For this load level (even with high-load transcoding), a multi-core machine certainly would not be needed. That said, it certainly wouldn't hurt anything to add on extra cores, especially if they're free ;-) -erik ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
I had the same problem with 45 polycom 601 phones in the same page group. It was just like you describe it and I got the same answer from polycom. What I did to go around that was add a second line key with a different extension number on each phone and then create the page group with the second extensions as members instead of the first extension. For example if one phone was extension 154, I added a second line key assigned extension 8154 to that phone only for the purpose of receiving pages and then added 8154 to the page group instead of 154. This way presence can still be enabled for 154 while 8154 which is not watched (and therefore will not send out those presence notifications which cause the phone to reboot) can be used to receive the pages. Joseph From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Al lists Sent: Wednesday, October 10, 2007 1:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Paging in Asterisk i'm using Polycom 601 in an office of 30 handsets. I have not heard my customer complaining about phones being rebooted after page. On 10/9/07, Bill Andersen [EMAIL PROTECTED] wrote: I could not tell you in asterisknow but I use this feature with Polycom phones on all of my installs. It is very well documented in voip-info.org Do you have any problem with the Paging when there are say 20 phones in the page group? We have a IP601 that is used by the receptionist and has 2 side cars. We have to keep presence (Buddy List) enabled so the sidecar lights go on and off. However, about 1 out of 10 times the receptionist pages, her phone reboots. Polycom says it can't handle the traffic from the buddy list presence notifications. Have you seen this? Bill ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging possible on an ATA?
At 23:41 10/10/2007, Luki wrote: Is it possible to configure a PAP2 to auto-answer for either paging or intercom? No. You cannot force the connected device (phone) to auto-answer. Imagine you have a plain old phone attached to it, who's going to lift the receiver? Yep, I guess even if the ATA could simulate the change from 3 MOhms to 600 Ohms, the phone needs to flip a switch internally. Oh well. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Mask Initial Processing with Ring Back Tone
I need to process a number of lines of code in the dialplan before answering a call. Can standard ring back tones be played to the caller while this is happening prior to answering the call. Which commands would facilitate this? Thanks in Advance, Vic ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
Erik Anderson wrote: For this load level (even with high-load transcoding), a multi-core machine certainly would not be needed. That said, it certainly wouldn't hurt anything to add on extra cores, especially if they're free ;-) Raul, The points concerning overall load are valid, but I agree with Erik's statement about getting the extra cores if they are free. Asterisk is heavily multi-threaded (one thread per channel plus several core threads), so a system with 35 simultaneous calls will happily balance the load across 8 (or more) cores. Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mask Initial Processing with Ring Back Tone
On 10/11/07, James FitzGibbon [EMAIL PROTECTED] wrote: What you do in between is up to you. Many people use something like Wait(2) to give a comfort ring, since PRI-connected incoming calls can often be set up nearly instantaneously. You'd want to limit the time obviously, and have proper exception handling in case whatever you're doing between Ringing() and Answer() fails. I should add that some applications (usually ones dealing with audio) will answer the channel for you, so you do have to be cognizant of that. If what you're doing in between is information processing, you should be OK, but I believe calling an AGI will auto-answer the channel, limiting what you can do somewhat. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mask Initial Processing with Ring Back Tone
On 10/11/07, Victor [EMAIL PROTECTED] wrote: I need to process a number of lines of code in the dialplan before answering a call. Can standard ring back tones be played to the caller while this is happening prior to answering the call. Which commands would facilitate this? You start sending ringback with Ringing() You answer with Answer() What you do in between is up to you. Many people use something like Wait(2) to give a comfort ring, since PRI-connected incoming calls can often be set up nearly instantaneously. You'd want to limit the time obviously, and have proper exception handling in case whatever you're doing between Ringing() and Answer() fails. -- j. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging possible on an ATA?
Doug wrote: At 23:41 10/10/2007, Luki wrote: Is it possible to configure a PAP2 to auto-answer for either paging or intercom? No. You cannot force the connected device (phone) to auto-answer. Imagine you have a plain old phone attached to it, who's going to lift the receiver? Yep, I guess even if the ATA could simulate the change from 3 MOhms to 600 Ohms, the phone needs to flip a switch internally. Oh well. There are, or were, Telecom devices that would connect to a station line, auto answer and bridge the audio to a paging amplifier with configurable timeout that would do what you want. ValCom and Viking come to mind. John Novack -- Dog is my co-pilot ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk System Setup Question
Hi All, I have done some research on Asterisk and I would like to try it in my office. Here's what I'm looking at for my system: AsteriskNow running one of two ways: 1) As a virtual machine on a VMWare server (Eight core Xeon server with 4GB ram) 2) on a P4 2.4Ghz with 768mb RAM I'm looking at 4-5 phones in the office. I was going to go with Grandstream or Polycom phones but I've heard people have had some issues with that. The Aastra 51i seems to be a good choice. I'd eventually like to link my Asterisk installation with my CRM, so the XML capabilities of the 51i should come in handy. I'm going to use SIP Trunking through Bandwidth.com with 2-3 SIP trunks. I'd like to know what more experienced users think of this setup, and if anyone has had any experience with either the 51i phones or the Bandwidth.com service. Also, has anyone had problems using a fax machine (either computer or paper) with Asterisk? Thank you in advance for your input! Regards, Zaheer K. Master President, Adamant Security Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mask Initial Processing with Ring Back Tone
Victor wrote: I need to process a number of lines of code in the dialplan before answering a call. Can standard ring back tones be played to the caller while this is happening prior to answering the call. Which commands would facilitate this? I strongly doubt those lines are going to take up much time. You can use Playtones to play specific inband tones. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mask Initial Processing with Ring Back Tone
Just dont answer it till the processing is done. No debate is needed for this. I do this millions of times per month. /b On Oct 11, 2007, at 2:56 PM, Eric \ManxPower\ Wieling [EMAIL PROTECTED] wrote: Victor wrote: I need to process a number of lines of code in the dialplan before answering a call. Can standard ring back tones be played to the caller while this is happening prior to answering the call. Which commands would facilitate this? I strongly doubt those lines are going to take up much time. You can use Playtones to play specific inband tones. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Paging in Asterisk
Joseph Begumisa wrote: I had the same problem with 45 polycom 601 phones in the same page group. It was just like you describe it and I got the same answer from polycom. What I did to go around that was add a second line key with a different extension number on each phone and then create the page group with the second extensions as members instead of the first extension. Interesting. I've seen the reboot mystery mentioned before. Some have pointed to power, but this makes sense. Do you know if this is still an issue on the newer series (330,550,650) phones as well? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Mask Initial Processing with Ring Back Tone
Brian West wrote: Just dont answer it till the processing is done. No debate is needed for this. I do this millions of times per month. Yes, this is one of those things too simple to be obvious. Like Brian said, just do your processing and THEN Answer() -- Generally, the caller will get ringback already, from their telco. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed FAX - How to best complement asterisk ?
Ex Vito wrote: 2. On the remaining locations we have a problem which I have been studying and trying to address... Faxing over IP. Could the 'remote' locations make do without a fax machine proper? We have sheet-fed pdf scanners here, drop the document in and hit the button, and acrobat shows up; hit print, select the printer called Fax, hit OK, and type in a phone number. Done. The last bit (the fake printer) is installed by WinPrintHylaFax [1] which is a windows client that sends jobs over IP to a hylafax server.I'm not sure how attached to a manual fax machine your users are, but mine sure were, and this sheet-fed pdf scanner combined with winprinthylafax appeased them. Moj [1] http://winprinthylafax.sourceforge.net/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Bridging in Asterisk
Am I correct in understanding that if the call comes in g729 and it is ended in g729 ( by the provider ) , asterisk does only bridging, therefore using very few CPU ressources ? Am I correct in understanding that this bridging means that calls ( rtp ) pass from one provider to another, therefore using low bandwith? Thank you A. Helping businesses save money worldwide www.sunapemobil.ca Numar de acces in .ro : +40318107430 (se taxeaza la pretul unui numar 031) Romania 3.5 c/min (USD) Moldova 10c/min - Looking for a deal? Find great prices on flights and hotels with Yahoo! FareChase.___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Opinions on Release Numbering
On Thursday 11 October 2007 12:45:45 Jay R. Ashworth wrote: People who know little should not be *trying* to interpret version numbers; they should be using what a packager, a website, or a knowledgeable other source *tells* them to use. This I disagree with, fundamentally. People should be able to pull themselves up by their bootstraps, if they choose to go down that road. I speak from personal experience, here. While I am certainly one of the core developers and fairly high up on the totem pole, I still remember my frustrations as a young programmer, and this is an attempt to take care of one of those frustrations. Yes, it seems so obvious *now*. Why can't we dump tradition and try to build a versioning system that IS more obvious? -- Tilghman ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] GTALK problem
Thank you I need to wait the international version of gtalk On 10/11/07, Philippe Sultan [EMAIL PROTECTED] wrote: If I calling asterisk with GTALK in english everything is ok, however, some of my friends with the italian version of gtalk they cannot have the audio. Audio problems might be experienced with older Gtalk clients. Version 1.0.0.104 is reported to work. The following resources may help you : http://www.voip-info.org/wiki/view/Asterisk+Google+Talk#Bugsampknownissues http://bugs.digium.com/view.php?id=10512 Hope this will help you solve the problem, Philippe ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Bridging in Asterisk
That is brought to you by the sip reinvite, in short yes, unless you set canreinvite = no to either side of that. Apa Minerala wrote: Am I correct in understanding that if the call comes in g729 and it is ended in g729 ( by the provider ) , asterisk does only bridging, therefore using very few CPU ressources ? Am I correct in understanding that this bridging means that calls ( rtp ) pass from one provider to another, therefore using low bandwith? Thank you A. Helping businesses save money worldwide www.sunapemobil.ca Numar de acces in .ro : +40318107430 (se taxeaza la pretul unui numar 031) Romania 3.5 c/min (USD) Moldova 10c/min Looking for a deal? Find great prices on flights and hotels http://us.rd.yahoo.com/evt=47094/*http://farechase.yahoo.com/;_ylc=X3oDMTFicDJoNDllBF9TAzk3NDA3NTg5BHBvcwMxMwRzZWMDZ3JvdXBzBHNsawNlbWFpbC1uY20- with Yahoo! FareChase. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thank you and have a wonderful day, Anthony Francis Rockynet VOIP ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] really sorry about this - E1 vs T1
I am *really* sorry about hijacking this thread, but the only way I can post to the -user list is by replying to another thread. (btw, this is getting really annoying. Please, Digium, sort the filters out!) I installed my super-duper new TE412P card today, without remembering to check the settings for T1/E1. As the server is now a hundred miles away, is there a) Any way of checking what setting is in place b) Changing that setting without having to physically remove the card and see ? Julian. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] aastra 9133i and autoanswer with headset
I am *really* sorry about hijacking this thread, but the only way I can post to the -user list is by replying to another thread. (btw, this is getting really annoying. Please, Digium, sort the filters out!) I've added the auto-answer header in my dialplan, and it works great. However, there is a problem with a headset - If I use a headset, then no matter what the settings on the phone are, the auto-answer *always* defaults to the speakerphone rather than the headset. Has anyone else encountered this, and have a solution ? Many thanks Julian ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is there real benefits on a SMP machine for Asterisk?
Hi Gerald, Well we have 2 APC UPSes in the server room, so each power supply will be connected to one UPS, and the UPSes are connected to (a transfer system of) an auxiliary power generator that start in less than a minute after a blackout. The server will have RAID5, of SAS disc But thanks for bringing up to my notice the needs for a quick replacement of my TDM hardware (FXO ports). On 10/11/07, Gerald A [EMAIL PROTECTED] wrote: Uptime is nice, but you want your system to be reliable when it's needed. Rather then buy a fancy fancy box, think carefully about what is precious and make sure that is robust. If you need your voicemail always available, think about some kind of mirror or RAID. If you are using TDM hardware (direct to some type of phone interface) check how long it will take to get a replacement, or keep one on warm standby. Redundant power supplies are fancy, but a good UPS is probably worth more. It's easy to buy a system with lots of bells and whistles -- just make sure they are the right kind. And your clunker might make a good lab/warm backup box. Hope this helps, Gerald. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] really sorry about this - E1 vs T1
Julian Lyndon-Smith wrote: I am *really* sorry about hijacking this thread, but the only way I can post to the -user list is by replying to another thread. (btw, this is getting really annoying. Please, Digium, sort the filters out!) I installed my super-duper new TE412P card today, without remembering to check the settings for T1/E1. As the server is now a hundred miles away, is there a) Any way of checking what setting is in place b) Changing that setting without having to physically remove the card and see ? Yes, see the t1e1override module parameter in wct4xxp/base.c. IIRC, it's 0xff to hard code to E1 mode, and set it to 0 for T1 mode. -1 is to use the jumper settings. -- Matthew Fredrickson Software/Firmware Engineer Digium, Inc. ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Distributed FAX - How to best complement asterisk ?
Mojo with Horan Company, LLC wrote: Ex Vito wrote: 2. On the remaining locations we have a problem which I have been studying and trying to address... Faxing over IP. Could the 'remote' locations make do without a fax machine proper? We have sheet-fed pdf scanners here, drop the document in and hit the button, and acrobat shows up; hit print, select the printer called Fax, hit OK, and type in a phone number. Done. The last bit (the fake printer) is installed by WinPrintHylaFax [1] which is a windows client that sends jobs over IP to a hylafax server.I'm not sure how attached to a manual fax machine your users are, but mine sure were, and this sheet-fed pdf scanner combined with winprinthylafax appeased them. Moj [1] http://winprinthylafax.sourceforge.net/ ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users We are faxing over SIP trunks from bandwidth.com and have 5 fax numbers all working without any problem. They are iaxmodem + hylafax. We can also send and receive faxes with tx_fax app. The big key is to have a bandwidth manager between your internet connection and your servers. Jonn ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] MINNESOTA: TwinCities Asterisk Users Group Meeting - October 13th 2007 11:30AM
Meeting Start: 10/13/2007 - 11:30am Hello all Twin Cities Asterisk Users, It's time once again to have another meeting. We'll talk about what's new with the Digium product line, discuss everything you n/ever wanted to know about dtmf and how each channel is configured and interoperates as far as touch-tones(r) go. I'd also like to hear from anyone who attended Astricon. This is the first one I've missed. I think we may have a chance to review an inexpensive meshing wireless product too. I'd like to thank O'reilly for the books we'll be giving away at this meeting. Their support has been fanstic! Be sure to get here early as the chairs go fast. The October 13th meeting will be starting at 11:30am. 7839 12th Ave S. Bloomington MN 55425 To Everyone, a very special THANK YOU! Eric Osterberg, Sound Choice Communications LLC (651)-999-0888 - Voice Line PS: * Asterisk is a registered trademark of Digium. With great respect we use the name. PPS: Touch-tone is a registered trademark of somebody too... What's their name again? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] German SIP and/or IAX providers?
Hi, all. My company is setting up a branch office in Germany, and I'm very interested in a VoIP provider over thataway. However, I'd need a few things: - Reliability. Can't have my branch office's DID's just going down. A company with a proven track record would be very, very good. - English. I speak great English, decent Spanish... and zilch German. So, as provincial as it might make me, I need a company I could talk to if the chips are down. And that's about -it-. I'm even willing to pay a reasonable premium, so long as it gets me a VoIP provider with the above restrictions. Any suggestions? Thanks much! -Ken ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Difference between trunk and released versions
Hello Sean, Does this clear things up? Yes! Thanks! __Yehavi: ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] HOw to call queue ???
HOw to call queues in asterisk ? ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
