Re: [asterisk-users] Help on g729 CODEC
Hi All, I need a help on g729 codec.Is there any tool which can convert g711 codec into g729 codec and supports batch processing ? Thanks in advance vivek --- On Fri, 11/7/08, Edgar Guadamuz [EMAIL PROTECTED] wrote: From: Edgar Guadamuz [EMAIL PROTECTED] Subject: Re: [asterisk-users] asterisk and bigmem kernel To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Date: Friday, November 7, 2008, 9:45 AM What happens?... I'm not sure. IAX peers work fine, but SIP users does not register. There are not firewalls blocking ports. But actually the problem is not the issue because I tried with normal kernel and doesn't work. It is not configuration because it worked on a virtual machine on VirtualBox. Even more strange, Trixbox DOES work. I think I'll continue with trixbox by the moment On Sun, Nov 2, 2008 at 11:18 PM, Tzafrir Cohen [EMAIL PROTECTED]wrote: On Sun, Nov 02, 2008 at 10:03:12PM -0600, Edgar Guadamuz wrote: Hi all, I installed asterisk 1.4.22 on a Dell poweredge 2950, with 4GB RAM. I used debian, but the default kernel doesn't recognize the 4GB, just 3, so I installled the linux-image-2.6.18-6-686-bigmem kernel, that do recognize the whole 4GB. Asterisk seems to be installed correctly, but I had two issues: (1) I had an error with zaptel. Asterisk didn't start with zaptel modules loaded. I had to rmmod zaptel to get asterisk running. lsmod | grep ^zaptel zttest -c 3 (2) SIP doesn't work What did you do? What did you expect to happen? What actually happened? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][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 ___ -- 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] Asterisk Gtalk setup
Hi, I've just followed http://www.voip-info.org/wiki/view/Asterisk+Speaks+with+Google+Talkinstructions from wiki, And i always get my jabber (GoogleTalk account for asterisk server) not registred: asterisk1*CLI jabber show connected Jabber Users and their status: User: myasteriskaccount - Disconnected asterisk1*CLI jabber test ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Provider recommendation in USA
Hi, I would like to seek an opinion or list of providers in USA or particularly in California. We would need someone who can offer maximum ports and lowest rates. Thanks very much, Vivek ___ -- 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] Asterisk on Solaris
Hi, try adding this in your stdtime/localtime.c #define _POSIX_PTHREAD_SEMANTICS #undef TM_ZONE #undef TM_GMTOFF if this does not work just google it, there are workaround for this problem Thanks, Vivek On 12/2/07, Mike Clark [EMAIL PROTECTED] wrote: I submiited to the list last night, but it never showed up. Here we go again. I've tried building Asterisk 1.4.15 on Solaris based on instuctions here, http://forums.digium.com/viewtopic.php?t=5888. However, this is the message I get. This is Solaris on X86. Any ideas? [CC] stdtime/localtime.c - stdtime/localtime.o stdtime/localtime.c: In function `localsub': stdtime/localtime.c:1136: error: structure has no member named `tm_gmtoff' gmake[1]: *** [stdtime/localtime.o] Error 1 gmake: *** [main] Error 2 Thanks, Mike Clark ___ --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] How to originate a call from console CLI ?
I am not sure if this fits in your requirement but try dial command. --Vivek On 11/29/07, Olivier [EMAIL PROTECTED] wrote: Hi, I would like to originate my first call from CLI. As I'm new to this, I'm wondering if it's possible. When I type originate from CLI, I've got this : There are two ways to use this command. A call can be originated between a channel and a specific application, or between a channel and an extension in the dialplan. This is similar to call files or the manager originate action. Calls originated with this command are given a timeout of 30 seconds. Usage1: originate tech/data application appname [appdata] This will originate a call between the specified channel tech/data and the given application. Arguments to the application are optional. If the given arguments to the application include spaces, all of the arguments to the application need to be placed in quotation marks. Usage2: originate tech/data extension [EMAIL PROTECTED] This will originate a call between the specified channel tech/data and the given extension. If no context is specified, the 'default' context will be used. If no extension is given, the 's' extension will be used. I would like for example to call 0123456789 number from SIP/7530 extension. My asterisk server is set to use local context for outgoing calls. My first idea was to type this : originate SIP 7530 [EMAIL PROTECTED] But it fails : it keeps displaying There are two ways ... and nothing else seem to occur. Can anyone help ? Cheers ___ --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] Registration state: Failed
Hi, x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues. Thanks, Vivek On 11/30/07, Newbie [EMAIL PROTECTED] wrote: Dear Support, I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line. I have 3 extensions: 250 - my extension 998 - I configured as Line 1 in SPA-3102 999 - I configured as PSTN Line 1 in SPA-3102 I have created 998 and 999 to the user extension list of the AsteriskNow why I still got Registration state: Failed for both Line 1 status and PSTN Line status ? my topology is below: Users -- AsteriskNow -- SPA-3102 -- PSTN line Please help Thanks a lot in advance Regards Winanjaya ___ --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] Registration state: Failed
well, then i would recommend to see full log in debug mode that might give some clue. if you have not done this before you can uncomment line starting with full= in the logger.conf... the log will be the usual /var/log/asterisk/ directory. Thanks, Vivek On 11/30/07, Newbie [EMAIL PROTECTED] wrote: Hi, there is no problem with X-Lite, the problem is SPA-3102 shown: Line 1: Registration Status: Failed PSTN Line 1: Registration Status: Failed I also had added 1 more extension 251..then tried to call 251 from 250 by using X-Lite and it works perfectly.. so that's why I am sure there is no problem with X-Lite .. what I suspect is the problem on Registration process in AsteriskNow.. since I am very new with this.. I don't know why this problem occurs ... could any body please help? Thanks Regards Winanjaya [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes videosupport=yes disallow=all allow=ilbc allow=gsm allow=ulaw allow=h261 allow=h263 allow=h263p register=998:[EMAIL PROTECTED]/998 register=999:[EMAIL PROTECTED]/999 [line1] type=peer host=dynamic defaultip=172.16.1.74 fromuser=998 secret=1234 fromdomain=172.16.1.169 [line2] type=peer host=dynamic defaultip=172.16.1.74 username=999 secret=1234 fromdomain=172.16.1.169 Command* sip show peers* Name/username HostDyn Nat ACL Port Status pstnline1/999 (Unspecified)D 0Unmonitored line1 (Unspecified)D 0Unmonitored 250/250172.16.1.88 D 27778Unmonitored 2500 (Unspecified)D 0Unmonitored 251(Unspecified)D 0Unmonitored 5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline] - Original Message - *From:* Vivek Shrivastava [EMAIL PROTECTED] *To:* Newbie [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Sent:* Saturday, December 01, 2007 11:34 AM *Subject:* Re: [asterisk-users] Registration state: Failed Hi, x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues. Thanks, Vivek On 11/30/07, Newbie [EMAIL PROTECTED] wrote: Dear Support, I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line. I have 3 extensions: 250 - my extension 998 - I configured as Line 1 in SPA-3102 999 - I configured as PSTN Line 1 in SPA-3102 I have created 998 and 999 to the user extension list of the AsteriskNow why I still got Registration state: Failed for both Line 1 status and PSTN Line status ? my topology is below: Users -- AsteriskNow -- SPA-3102 -- PSTN line Please help Thanks a lot in advance Regards Winanjaya ___ --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] Registration state: Failed
Hmmm, what OS you are using,,,this could be related to *Access Control Lists..*but i guess that is in Solaris * * On 11/30/07, Newbie [EMAIL PROTECTED] wrote: Hello, After I turned on full= in logged.conf .. I got the following: [Nov 30 12:00:25] NOTICE[2601] chan_sip.c: Registration from 'FXS1 sip:[EMAIL PROTECTED]' failed for '172.16.1.169' - Device does not match ACL [Nov 30 12:00:29] NOTICE[2601] chan_sip.c: Registration from 'FXS1 sip:[EMAIL PROTECTED]' failed for '172.16.1.169' - Device does not match ACL [Nov 30 12:00:33] NOTICE[2601] chan_sip.c: Registration from 'FXS1 sip:[EMAIL PROTECTED]' failed for '172.16.1.169' - Device does not match ACL [Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 sip:[EMAIL PROTECTED]' failed for '172.16.1.169' - Device does not match ACL [Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 sip:[EMAIL PROTECTED]' failed for '172.16.1.169' - Device does not match ACL [Nov 30 12:00:36] NOTICE[2601] chan_sip.c: Registration from 'FXO1 sip:[EMAIL PROTECTED]' failed for '172.16.1.169' - Device does not match ACL any idea or clue? Thanks a lot in advance Regards Winanjaya - Original Message - *From:* Vivek Shrivastava [EMAIL PROTECTED] *To:* Newbie [EMAIL PROTECTED] *Cc:* Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com *Sent:* Saturday, December 01, 2007 11:50 AM *Subject:* Re: [asterisk-users] Registration state: Failed well, then i would recommend to see full log in debug mode that might give some clue. if you have not done this before you can uncomment line starting with full= in the logger.conf... the log will be the usual /var/log/asterisk/ directory. Thanks, Vivek On 11/30/07, Newbie [EMAIL PROTECTED] wrote: Hi, there is no problem with X-Lite, the problem is SPA-3102 shown: Line 1: Registration Status: Failed PSTN Line 1: Registration Status: Failed I also had added 1 more extension 251..then tried to call 251 from 250 by using X-Lite and it works perfectly.. so that's why I am sure there is no problem with X-Lite .. what I suspect is the problem on Registration process in AsteriskNow.. since I am very new with this.. I don't know why this problem occurs ... could any body please help? Thanks Regards Winanjaya [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes videosupport=yes disallow=all allow=ilbc allow=gsm allow=ulaw allow=h261 allow=h263 allow=h263p register=998:[EMAIL PROTECTED]/998 register=999:[EMAIL PROTECTED]/999 [line1] type=peer host=dynamic defaultip=172.16.1.74 fromuser=998 secret=1234 fromdomain=172.16.1.169 [line2] type=peer host=dynamic defaultip=172.16.1.74 username=999 secret=1234 fromdomain=172.16.1.169 Command* sip show peers* Name/username HostDyn Nat ACL Port Status pstnline1/999 (Unspecified)D 0Unmonitored line1 (Unspecified)D 0Unmonitored 250/250172.16.1.88 D 27778Unmonitored 2500 (Unspecified)D 0Unmonitored 251(Unspecified)D 0Unmonitored 5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline] - Original Message - *From:* Vivek Shrivastava [EMAIL PROTECTED] *To:* Newbie [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Sent:* Saturday, December 01, 2007 11:34 AM *Subject:* Re: [asterisk-users] Registration state: Failed Hi, x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues. Thanks, Vivek On 11/30/07, Newbie [EMAIL PROTECTED] wrote: Dear Support, I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line. I have 3 extensions: 250 - my extension 998 - I configured as Line 1 in SPA-3102 999 - I configured as PSTN Line 1 in SPA-3102 I have created 998 and 999 to the user extension list of the AsteriskNow why I still got Registration state: Failed for both Line 1 status and PSTN Line status ? my topology is below: Users -- AsteriskNow -- SPA-3102 -- PSTN line Please help Thanks a lot in advance Regards Winanjaya ___ --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
Re: [asterisk-users] Registration state: Failed
you can also look at this... http://www.asteriskguru.com/tutorials/idefisk_20_free.html I has this error initially with Asterisk server when I try to register. Device does not match ACL got it resolved by setting Caller ID Name : users exten On 11/30/07, Vivek Shrivastava [EMAIL PROTECTED] wrote: Hmmm, what OS you are using,,,this could be related to *Access Control Lists..*but i guess that is in Solaris * * On 11/30/07, Newbie [EMAIL PROTECTED] wrote: Hello, After I turned on full= in logged.conf .. I got the following: [Nov 30 12:00:25] NOTICE[2601] chan_sip.c: Registration from 'FXS1 sip:[EMAIL PROTECTED]' failed for ' 172.16.1.169' - Device does not match ACL [Nov 30 12:00:29] NOTICE[2601] chan_sip.c: Registration from 'FXS1 sip:[EMAIL PROTECTED]' failed for ' 172.16.1.169' - Device does not match ACL [Nov 30 12:00:33] NOTICE[2601] chan_sip.c: Registration from 'FXS1 sip:[EMAIL PROTECTED]' failed for '172.16.1.169' - Device does not match ACL [Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 sip:[EMAIL PROTECTED]' failed for ' 172.16.1.169' - Device does not match ACL [Nov 30 12:00:35] NOTICE[2601] chan_sip.c: Registration from 'FXO1 sip:[EMAIL PROTECTED]' failed for '172.16.1.169' - Device does not match ACL [Nov 30 12:00:36] NOTICE[2601] chan_sip.c: Registration from 'FXO1 sip:[EMAIL PROTECTED]' failed for ' 172.16.1.169' - Device does not match ACL any idea or clue? Thanks a lot in advance Regards Winanjaya - Original Message - *From:* Vivek Shrivastava [EMAIL PROTECTED] *To:* Newbie [EMAIL PROTECTED] *Cc:* Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Sent:* Saturday, December 01, 2007 11:50 AM *Subject:* Re: [asterisk-users] Registration state: Failed well, then i would recommend to see full log in debug mode that might give some clue. if you have not done this before you can uncomment line starting with full= in the logger.conf... the log will be the usual /var/log/asterisk/ directory. Thanks, Vivek On 11/30/07, Newbie [EMAIL PROTECTED] wrote: Hi, there is no problem with X-Lite, the problem is SPA-3102 shown: Line 1: Registration Status: Failed PSTN Line 1: Registration Status: Failed I also had added 1 more extension 251..then tried to call 251 from 250 by using X-Lite and it works perfectly.. so that's why I am sure there is no problem with X-Lite .. what I suspect is the problem on Registration process in AsteriskNow.. since I am very new with this.. I don't know why this problem occurs ... could any body please help? Thanks Regards Winanjaya [general] context=default allowoverlap=no bindport=5060 bindaddr=0.0.0.0 srvlookup=yes videosupport=yes disallow=all allow=ilbc allow=gsm allow=ulaw allow=h261 allow=h263 allow=h263p register=998:[EMAIL PROTECTED]/998 register=999:[EMAIL PROTECTED]/999 [line1] type=peer host=dynamic defaultip=172.16.1.74 fromuser=998 secret=1234 fromdomain=172.16.1.169 [line2] type=peer host=dynamic defaultip=172.16.1.74 username=999 secret=1234 fromdomain=172.16.1.169 Command* sip show peers* Name/username HostDyn Nat ACL Port Status pstnline1/999 (Unspecified)D 0 Unmonitored line1 (Unspecified)D 0 Unmonitored 250/250172.16.1.88 D 27778 Unmonitored 2500 (Unspecified)D 0 Unmonitored 251(Unspecified)D 0 Unmonitored 5 sip peers [Monitored: 0 online, 0 offline Unmonitored: 1 online, 4 offline] - Original Message - *From:* Vivek Shrivastava [EMAIL PROTECTED] *To:* Newbie [EMAIL PROTECTED] ; Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com *Sent:* Saturday, December 01, 2007 11:34 AM *Subject:* Re: [asterisk-users] Registration state: Failed Hi, x-lite has extensive debug facility you can turn that on in the advanced options, that probably will give better understanding as what is going on from x-lite side. i also have experienced the same but that involved firewall and NAT issues. Thanks, Vivek On 11/30/07, Newbie [EMAIL PROTECTED] wrote: Dear Support, I am running AsteriskNow + X-Lite as my SoftPhone and SPA-3102 connected with PSTN line. I have 3 extensions: 250 - my extension 998 - I configured as Line 1 in SPA-3102 999 - I configured as PSTN Line 1 in SPA-3102 I have created 998 and 999 to the user extension list of the AsteriskNow
Re: [asterisk-users] How to originate a call from console CLI ?
yup with chan_oss On 11/30/07, Olivier [EMAIL PROTECTED] wrote: 2007/11/30, Vivek Shrivastava [EMAIL PROTECTED]: I am not sure if this fits in your requirement but try dial command. Do you mean, dialing both extensions one after the other and then, bridge them ? Or do you mean using the asterisk Chan_OSS capabilities ? Cheers ___ --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] Simple Asterisk to Asterisk SIP Call Setup?
looks like something wrong with the dial plan in the extensions.conf.. i would recommend start debug on and see the content of full log may be that give some clue. Thanks, Vivek On 11/30/07, Russell Brown [EMAIL PROTECTED] wrote: I have two Asterisk systems that can route to each other via a VPN with firewalls disabled for testing purposes. Each Server can see (tested via nmap) UDP port 5060 on the other. So... I thought that I could simply use a Dial command in Server A's config to place a SIP call to Server B... but it doesn't seem to work. Server A (192.168.1.33) has: exten = *136,1,Dial(SIP/[EMAIL PROTECTED],30) but whenever a user on Server A dials '*136' the call doesn't complete and the CLI shows: Executing [EMAIL PROTECTED]:1] Dial(SIP/112-0071f650, SIP/[EMAIL PROTECTED]|30) in new stack -- Called [EMAIL PROTECTED] -- SIP/10.10.111.13-00793520 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) I can't see anything in Server B's logs from 192.168.1.33 What am I missing? Any pointers to help me get this working? -- Regards, Russell | Russell Brown | MAIL: [EMAIL PROTECTED] PHONE: 01780 471800 | | Lady Lodge Systems | WWW Work: http://www.lls.com | | Peterborough, England | WWW Play: http://www.ruffle.me.uk | ___ --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] Newb Question
you can try Cain Abel ( to route calls) and Wireshark to record all the calls. On 11/29/07, Adam Moffett [EMAIL PROTECTED] wrote: I'm pretty sure asterisk won't do that without modification. You'll need to do packet sniffing and decode the datathere may be products that do this, but asterisk is not it. And we're assuming the calls are unencrypted? I inherited an office with phones that are hosted off-site. Everything is skinny and G729. I see that the FreeBSD asterisk port comes with a G729 codec. I want to record everything. If I use port mirroring on my switch, is it possible to configure asterisk to record and assemble packets that it doesn't otherwise route? Is it insane to user asterisk for this purpose? Advice or a link to a howto would be greatly appreciated. ___ --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] Asterisk-Users: Termination
We are using only voip chanels with 400-500 channels. Although we are still in begining phase but i have not seen any problem as such. Thanks, Vivek On 11/20/07, Mark Adams [EMAIL PROTECTED] wrote: I wanted to see if anyone has set up a large amount of out bound only voip channels? We run analog autodialers connected to analog to voip gateways (dialogic boards to audiocodes mp-124's) Bandwidth and routing is being provided by a bonded t-1 (3 meg) cisco 2600 with 2 t1 cards and a 16 port netgear switch. My question ( if you can picture the setup) is if anyone can see a problem with the set-up I have described. There is no firewall or access list on the router. Just wide open internet. I have been running about 80 channels for over a year and my numbers have been down and I cannot tell if there are any problems Mark Adams Infinity Marketing Inc. 1-800-430-1478 Main 530-579-8856 Fax 216-441-4319 Tech Support ___ --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] RTP traffic not being forwarded
Hi Ryan, Are the SIP and RTP ports are randomly selected or there are specific ports for these? Unchecking random port selection option on the device/softphone may help. --Vivek On 11/10/07, Ryan Newington [EMAIL PROTECTED] wrote: Hi Luki, Thanks for your advice. I've checked the firewall and it is set to allow all incoming traffic. I changed the media port range as well with no success. Some calls work fine. This is the configuration that doesn't work. The RTP traffic passes along the chain fine, but the Asterisk server doesn't do anything with the packets it gets from the near-end SIP phone and the media gateway. SIP Phone - Media Gateway - Asterisk - SIP Phone An asterisk internal call will work fine. Eg; SIP Phone - Asterisk - SIP Phone Regards Ryan -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] On Behalf Of Luki Sent: Sunday, 11 November 2007 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded When using 'rtp debug' on the asterisk console, it shows that it is receiving traffic from one endpoint, but not the other. A wireshark trace reveals it is actually receiving traffic from both ends. Sounds like a firewall issue. Wireshark shows what's on the wire, i.e. before iptables. The packets are being dropped for whatever reason and never reach the asterisk process. Check your iptables and RTP port range, and perhaps try changing it. Luki ___ --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 ___ --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] RTP traffic not being forwarded
Hi Ryan, I was just wondering if they need to be according rtp.conf. ( or you may need to modify rtp.conf) Regards, Vivek On 11/11/07, Ryan Newington [EMAIL PROTECTED] wrote: Hi Vivek, The SIP port is set to the standard port 5060. The RTP ports as far as I know are random ephemeral ports between 63000 and 64000. I can change the port range on the media server, asterisk and the device, but neither seems to help. My diagram below is probably misleading. The RTP traffic flow that I see is as follows (one way traffic into Asterisk) SIP Phone --- Media Gateway *---* Asterisk *---* SIP Phone Ryan *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Vivek Shrivastava *Sent:* Sunday, 11 November 2007 5:19 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] RTP traffic not being forwarded Hi Ryan, Are the SIP and RTP ports are randomly selected or there are specific ports for these? Unchecking random port selection option on the device/softphone may help. --Vivek On 11/10/07, *Ryan Newington* [EMAIL PROTECTED] wrote: Hi Luki, Thanks for your advice. I've checked the firewall and it is set to allow all incoming traffic. I changed the media port range as well with no success. Some calls work fine. This is the configuration that doesn't work. The RTP traffic passes along the chain fine, but the Asterisk server doesn't do anything with the packets it gets from the near-end SIP phone and the media gateway. SIP Phone - Media Gateway - Asterisk - SIP Phone An asterisk internal call will work fine. Eg; SIP Phone - Asterisk - SIP Phone Regards Ryan -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Luki Sent: Sunday, 11 November 2007 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded When using 'rtp debug' on the asterisk console, it shows that it is receiving traffic from one endpoint, but not the other. A wireshark trace reveals it is actually receiving traffic from both ends. Sounds like a firewall issue. Wireshark shows what's on the wire, i.e. before iptables. The packets are being dropped for whatever reason and never reach the asterisk process. Check your iptables and RTP port range, and perhaps try changing it. Luki ___ --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 ___ --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] IMAP Voicemail -- HELP! Asterisk not playing Greeting!
I would recommed to convert that to gsm format http://www.voip-info.org/tiki-index.php?page=Convert+WAV+audio+files+for+use+in+Asterisk On 11/11/07, Michael Schwartz [EMAIL PROTECTED] wrote: I'm using Asterisk 1.4.13, the latest released version. The linux platform is FC7. I setup my Asterisk server to use IMAP storage. Dovecot is the IMAP server. Its storing messages perfectly--no problems. I should also mention that I'm using MySQL for real-time configuration. That must be working (at least partially), because I can authenticate v. the voicemail table. However, the Voicemail system does not play a custom unavailable greeting. When I record the greeting (using the std voicemail method), it creates a valid .WAV file, which I can play using the Playback command. However, when an incoming voicemail is received, Asterisk plays the default system greeting. I see this warning in the logs: //...2125551212 is the sample voicemail box in this example...// [Nov 11 12:30:00] WARNING[8343] app_voicemail.c: Failed to open file: /var/spool/asterisk/voicemail/consumer/2125551212/unavail/msg-001.WAV: No such file or directory I'm not sure if this is actually bad though, because it is recording the messages in /var/spool/asterisk/voicemail/consumer/2125551212. Any ideas? I don't understand why the voicemail system would know how to record the greeting, but not play it. Thanks in advance! I'm really stumped on this one. - Mike Schwartz ___ --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] RTP traffic not being forwarded
well i think rtp port range is defined in rtp.conf and correct me if i am wrong, these ports must be opened/forwarded to communicate. http://www.asteriskguru.com/tutorials/sip_nat_oneway_or_no_audio_asterisk.html Let me know if you need more information. Thanks, Vivek On 11/11/07, Ryan Newington [EMAIL PROTECTED] wrote: Hi Vivek, I'm not sure what you mean, could you explain further? Regards Ryan *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Vivek Shrivastava *Sent:* Monday, 12 November 2007 1:21 AM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] RTP traffic not being forwarded Hi Ryan, I was just wondering if they need to be according rtp.conf. ( or you may need to modify rtp.conf) Regards, Vivek On 11/11/07, *Ryan Newington* [EMAIL PROTECTED] wrote: Hi Vivek, The SIP port is set to the standard port 5060. The RTP ports as far as I know are random ephemeral ports between 63000 and 64000. I can change the port range on the media server, asterisk and the device, but neither seems to help. My diagram below is probably misleading. The RTP traffic flow that I see is as follows (one way traffic into Asterisk) SIP Phone --- Media Gateway *---* Asterisk *---* SIP Phone Ryan *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Vivek Shrivastava *Sent:* Sunday, 11 November 2007 5:19 PM *To:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] RTP traffic not being forwarded Hi Ryan, Are the SIP and RTP ports are randomly selected or there are specific ports for these? Unchecking random port selection option on the device/softphone may help. --Vivek On 11/10/07, *Ryan Newington* [EMAIL PROTECTED] wrote: Hi Luki, Thanks for your advice. I've checked the firewall and it is set to allow all incoming traffic. I changed the media port range as well with no success. Some calls work fine. This is the configuration that doesn't work. The RTP traffic passes along the chain fine, but the Asterisk server doesn't do anything with the packets it gets from the near-end SIP phone and the media gateway. SIP Phone - Media Gateway - Asterisk - SIP Phone An asterisk internal call will work fine. Eg; SIP Phone - Asterisk - SIP Phone Regards Ryan -Original Message- From: [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] ] On Behalf Of Luki Sent: Sunday, 11 November 2007 12:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] RTP traffic not being forwarded When using 'rtp debug' on the asterisk console, it shows that it is receiving traffic from one endpoint, but not the other. A wireshark trace reveals it is actually receiving traffic from both ends. Sounds like a firewall issue. Wireshark shows what's on the wire, i.e. before iptables. The packets are being dropped for whatever reason and never reach the asterisk process. Check your iptables and RTP port range, and perhaps try changing it. Luki ___ --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 ___ --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 ___ --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] Wanted: tutorial on troubleshooting SIP issues
if your router and UA have syslog facility you can use RouterSyslog also. You can use Cain and Able with wireshark for switched network. Thanks, Vivek On 11/9/07, Alan Lord [EMAIL PROTECTED] wrote: Steve Edwards wrote: snip / Examples of what I'd like to see: 1) A SIP telephone registering successfully. 2) A SIP telephone failing to register for reasons x, y, and z. snip / I'm sorry but I don't see this as being very hard. Just install Wireshark and do it yourself... Alan ___ --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] Everyone is busy/congested: IP Trunk
Well, unfortunately i did not dig much into why/how it worked with openvpn, but it did work for me with default setup.I think you may need to set constant ports instead of random ports. Thanks, Vivek On 11/9/07, bilal ghayyad [EMAIL PROTECTED] wrote: Hi Friends; Actually I would appreciate if Vivek can advise if the VPN resolved the RTP packets in the SIP Trunk between Asterisk and another softswitch? In other words, openvpn helpful in NAT cases in what exactly? As without VPN, I was able to establish a call but without voice or with complete noise (nothing understood) :) - So if NAT resolve this issue for the SIP Trunk, then I can proceed forward, as really now I do not have any other attempt to try. From the other side, I think that baji is talking about something else than the IP Trunk, he is talking about outbound (which is related to using an application to run an outside call, which is used usually in campaign in contact centers and so on), I think nthis case differs that placing a calls via IP Trunk or even outside call but the caller who will do it (and not the application). Lastly, Mr. Amit helped me when he gave me a configuration to be done for the SIP Trunk, as in his method, I did not register on the softswitch, I send directly, and the connectioned succeed, but as I said: with complete voice (actually nothing understood, i feel it is complete RTP situation), the test was by letting Asterisk behind NAT (private IP) and sending to a softswitch in anther country has a public IP address. Is it NAT issue, so VPN can resolve? Note: anyone knows if h323 works better in the IP trunk? Regards Bilal -- yeah i found openvpn helpful in NAT cases. -Vivek On 11/6/07, Baji Panchumarti [EMAIL PROTECTED] wrote: after a copious loss of follicles :-), I finally got outbound working. Basically the channel statement in the call file needs to have the number to be called. For eg., in test.call format the statement as follows : Channel: SIP/3012345678@your-sip-provider And there is no need for a DIAL statement in extensions.conf unless you need to dial an additional number / extension. Then in sip.conf you need a para that matches your-sip-provider with the relevant auth info. These two wiki pages, they were very helpful in figuring out a solution to the problem : http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message hth, -baji. -- On Oct 30, 2007 8:43 AM, Gabriel Natale wrote: I have the same problem. I trying with more 4 SIP providers, the account is registering, receive inboud calls, but can`t make outbound calls for congestion. Can be the out call id the problem? Thanks Gabriel __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.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] 'a' extension
I think you can save/get the number in variable and then assign it to callerid. I am doing similar and working for me. Thanks, Viv On 11/8/07, Peder @ NetworkOblivion [EMAIL PROTECTED] wrote: Is there any way to see the called number when a call gets redirected to the 'a' extension from voicemail? Say x123 calls x456 and it rolls to voicemail. x123 hits * and gets dumped into the 'a' extension in the original context. I need some logic in 'a' to do a database lookup based on the original called number (x456). Any ideas? When I do a test, it appears that the called number is 'a' and the calling number is 123. I need to be able to tell that it was a call to x456. Thanks. Peder ___ --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] Asterisk and Grandstream both behind different NAT
Hi, i am facing some problem configuring 2 Grandstrem phones 101 behind NAT. I have put server ip as 192.x.x.x and OutGoing proxy as 72.x.x.x, i have forwarded ports on both Grandstream and Asterisk sides, and using those ports on Grandstream for SIP and RTP with random ports =no. This setup is working however at a time only one phone gets registered. Has someone experienced the same problemany suggestions? Thanks in advance, Viv ___ --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] Everyone is busy/congested: IP Trunk
yeah i found openvpn helpful in NAT cases. -Vivek On 11/6/07, Baji Panchumarti [EMAIL PROTECTED] wrote: after a copious loss of follicles :-), I finally got outbound working. Basically the channel statement in the call file needs to have the number to be called. For eg., in test.call format the statement as follows : Channel: SIP/3012345678@your-sip-provider And there is no need for a DIAL statement in extensions.conf unless you need to dial an additional number / extension. Then in sip.conf you need a para that matches your-sip-provider with the relevant auth info. These two wiki pages, they were very helpful in figuring out a solution to the problem : http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out http://www.voip-info.org/wiki/index.php?page=Asterisk+auto-dial+out+deliver+message hth, -baji. -- On Oct 30, 2007 8:43 AM, Gabriel Natale wrote: I have the same problem. I trying with more 4 SIP providers, the account is registering, receive inboud calls, but can`t make outbound calls for congestion. Can be the out call id the problem? Thanks Gabriel ___ --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] T.38 Faxing and Asterisk
Hi, Yes, i have used it for T.38 faxing. Thanks, Vivek On 10/26/07, Nasir Iqbal [EMAIL PROTECTED] wrote: Hi, Have you tried Callweaver http://www.callweaver.org ___ --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] help. newbie asterisk installation problem.
Hello friends, I am trying to install asterisk 1.4.0 . I am configuring it as follows:- ./configure --prefix=/home/vivek/downloads/install/asterisk/ But still while running 'make install', it tries to install it in /var/lib/asterisk/ and stops because of failing permissions. I have provided it a prefix, But it doesn't install it there. Can anybody tell me the solution for this. I dont want to install it in the default directories. I want it to be in my home directory. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon Electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --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] newbie asterisk 1.4 installation problem
Hello friends, I am trying to install asterisk 1.4. I am configuring it as follows:- ./configure --prefix=/home/vivek/downloads/install/asterisk/ But still while running 'make install', it tries to install it in /var/lib/asterisk/ and stops because of failing permissions. I have provided it a prefix, But it doesn't install it there. Can anybody tell me the solution for this. I dont want to install it in the default directories. I want it to be in my home directory. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon Electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --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] newbie astdb error, please help
I am getting this warning:- Oct 23 15:47:22 WARNING[2124]: db.c:171 ast_db_put: Unable to put value ' 192.168.1.12:5060:300:15553695861:sip:[EMAIL PROTECTED]:5060' for key '23' in family 'SIP/Registry I checked the file permissions. They are proper. There doesnot seem to be a visible error. No change has been done in any conf files for the past 4 months. The reinvite has also stopped. I dont have any idea whats happening. ___ --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] astdb error, please help
Hello friends, I am getting this error:- Oct 23 15:47:22 WARNING[2124]: db.c:171 ast_db_put: Unable to put value '192.168.1.12:5060:300:15553695861:sip:[EMAIL PROTECTED]:5060' for key '23' in family 'SIP/Registry I have no idea what it means. Please tell me what could be the problem. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --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] astdb error, please help
I checked the file permissions. They are proper. There doesnot seem to be a visible error. No change has been done in any conf files for the past 4 months. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ram wrote: check database On 10/23/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello friends, I am getting this error:- Oct 23 15:47:22 WARNING[2124]: db.c:171 ast_db_put: Unable to put value ' 192.168.1.12:5060:300:15553695861:sip:[EMAIL PROTECTED]:5060' for key '23' in family 'SIP/Registry I have no idea what it means. Please tell me what could be the problem. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --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] dialplan help
Dear friends,
Does anyone know how do i convert hex to int in the dialplan. I want to do
this:-
Take the sip call-id in hex, use CUT to extract the first part , and convert it
to an int. But the math function ony takes arguments as int. Can anyone suggest
how to do that?
eg:-
exten = _X.,n,Set(sipcid = ${CUT(SIPCALLID,-,1)}) --- evaluates to E305CEC5
I want this hex value in int. But i cant think of a clean solution.
Please help.
Thanks in advance.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
-- Ernest Rutherford
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Re: [asterisk-users] dialplan help
Hi Michael,
Thanks a lot. I am working on an agi script and it does it. Thanks a lot again.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
All science is either physics or stamp collecting.
-- Ernest Rutherford
Michiel van Baak wrote:
On 14:23, Wed 30 Aug 06, [EMAIL PROTECTED] wrote:
Dear friends,
Does anyone know how do i convert hex to int in the dialplan. I want to do
this:-
Take the sip call-id in hex, use CUT to extract the first part , and convert
it to an int. But the math function ony takes arguments as int. Can anyone
suggest how to do that?
eg:-
exten = _X.,n,Set(sipcid = ${CUT(SIPCALLID,-,1)}) --- evaluates to
E305CEC5
I want this hex value in int. But i cant think of a clean solution.
Please help.
Use a simple agi script that does this for you.
--
Michiel van Baak
[EMAIL PROTECTED]
http://michiel.vanbaak.eu
GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x71C946BD
Why is it drug addicts and computer afficionados are both called users?
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[asterisk-users] sip giving problems, please help.
sipsock_read: SIP MESSAGE JUST IGNORED: BYE Aug 29 15:10:29 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:29 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:29 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0 Aug 29 15:10:29 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:33 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:33 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: BYE Aug 29 15:10:33 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:33 ERROR[30036]: chan_sip.c:11323 sipsock_read: We could NOT get the channel lock for SIP/15552830438-990b! Aug 29 15:10:33 ERROR[30036]: chan_sip.c:11324 sipsock_read: SIP MESSAGE JUST IGNORED: SIP/2.0 Aug 29 15:10:33 ERROR[30036]: chan_sip.c:11325 sipsock_read: BAD! BAD! BAD! Aug 29 15:10:34 WARNING[30029]: channel.c:787 channel_find_locked: Avoided initial deadlock for '0x81bbd78', 10 retries! With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --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 dosenot compile
Hello friends, I am trying to install asterisk. I downloaded the latest development branch from digium thru svn. I get an error in the make which says:- [LD] codec_gsm.o gsm/lib/libgsm.a - codec_gsm.so [CC] codec_ilbc.c - codec_ilbc.o make[2]: Entering directory `/home/install/asterisk/codecs/ilbc' make[2]: *** virtual memory exhausted. Stop. make[2]: Leaving directory `/home/install/asterisk/codecs/ilbc' make[1]: *** [ilbc/libilbc.a] Error 2 make[1]: Leaving directory `/home/install/asterisk/codecs' make: *** [codecs] Error 2 It seems the codec ilibc has a problem. I tried touch the file ilbc.o and ilbc.so. But it wouldnot help. Please suggest how do I go further with this? Thankyou all in advance. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --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 gui
Hello friends, does anyone know if there is a gui for asterisk provided with the asterisk source or has to downloaded from somewhere else. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --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 download
Hi all, How do I download the development branch of asterisk 1.4. I am eagerly waiting for it. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --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] app background
Hello friends, I want to use the background(playfile) application without the channel being answered. I dont want playback because I would like the callee to dial the number while the file is being played. but I dont know how do i do that. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --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] audiocodes with asterisk:- newbie
Hello friends, I am using SIP on Asterisk 1.2.4. All my configurations are working perfectly on a Welltech fxo box. But today I changed to an audiocodes MP104 fxo box. All the sip signalling works fine but the noise is something like an alien invasion, I mean, its completely outrageous. I dont know what to do. Has anyone got an audiocodes with asterisk working. Please help me with some configurations in audiocodes With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. All science is either physics or stamp collecting. -- Ernest Rutherford ___ --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] Two asterisks on one machine
Hello friends, Can I run two asterisks running simultaneously on the same machine? I want one to run v1.0.2 for h323 ( which is an old and running production system ) and one for sip implementation. I wonder how it can be done since they will want access to the same ports and ip addresses. Does anyone know to do this or has done this before? Please share your experiences please. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --New opinions often appear first as jokes and fancies, then as blasphemies and treason, then as questions open to discussion, and finally as established truths. ___ --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] Two asterisks on one machine
Hi friend, I am running asterisk in production and it is being used by many people using h323. I cannot afford to change all their configurations. Also, the newer asterisk dosenot support inband for h323 properly. Thats why I want two asterisks one for backward compatibility and one for sip which I want to implement. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --New opinions often appear first as jokes and fancies, then as blasphemies and treason, then as questions open to discussion, and finally as established truths. Joseph Tanner wrote: You could run a virtual machine. I'd try xen, uml, and vmware in that order (vmware would be the easiest/quickest to setup, but is more of a resource-hog than xen or uml). Assign a separate ip to the virtual server, setup asterisk, and you're all set. BTW, just curious but why can't you run one asterisk install with both h323 and sip? It'd simplify things and use less resources than running a virtual server, assuming it works for you. Another idea, if one's solely for h323 and the other's solely for sip (neither will be running both), then you could compile asterisk twice, using different directories for each install. I don't think this would work if both needed to use the same ports. I'm guessing you want to bridge the h323 asterisk to the sip asterisk? If not, but you do want to use sip on both, perhaps you can use port 5060 on one and 5061 for the other. Couldn't bridge them, but both could talk to the outside world (that is, maybe they could, I haven't tried this and do not know what's involved). Running one in a virtual server is probably going to be the easiest way to get two asterisk processes to coexist on the same physical server. Joseph Tanner On 3/6/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: Hello friends, Can I run two asterisks running simultaneously on the same machine? I want one to run v1.0.2 for h323 ( which is an old and running production system ) and one for sip implementation. I wonder how it can be done since they will want access to the same ports and ip addresses. Does anyone know to do this or has done this before? Please share your experiences please. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --New opinions often appear first as jokes and fancies, then as blasphemies and treason, then as questions open to discussion, and finally as established truths. ___ --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] new jitter implementation for sip
Hello friends, I read in the jitter document in the asterisk-1.2.0/asterisk/doc/README.jitterbuffer that a new jitter implementation for SIP/RTP protocol is comming up. Does anyone know whether it has come up or when is it comming? It will be a great relief for all sip users. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Sweat saves blood, blood saves lives, and brains saves both. ___ --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] Rtp packets being dropped
Hello Friends, I am experiencing a problem. The rtp packets which detect dtmf from inband are being dropped. I have tried a priority ip address which allows voip packets first but it didnt work out. Asterisk is dropping only dtmf packets. I am using Sip protocol. Is there any way in asterisk whereby I can detect the dropped packets or enable their queueing or buffering? Please help, I am running out of ideas. Thanking you all. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Sweat saves blood, blood saves lives, and brains saves both. ___ --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] return code from AGI
Hello friends, Asterisk applications like Dial and other commands return codes. When AGI script is executed, it returns -1 on hangup and 0 on non hangup exit. How do I check these return codes from the extensions.conf . I want to check these return codes and control the dialplan. Please help me how do I track this. Thanks all for reading this mail. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Sweat saves blood, blood saves lives, and brains saves both. ___ --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] newbie dial problem,
Hello friends, I am using asterisk with sip phones and sip fxo box. My problem is that my dtmf is recognised internally only if I use dtmf=inband and outside to the pstn lines work only if I use dtmf=info. The result is that I cant transfer any calls from and to pstn. How do I fix this. Either one works properly or the other but not both of them. So when I have configured my boxes as dtmf=inband and I dial them inhouse. When I have to make a call outside, I say SipDtmfMode=info and dial outside. But then it doesnot transfer the call. Please help me. I am stuck up. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Sweat saves blood, blood saves lives, and brains saves both. ___ --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] Custom cdr trouble, help this newbie
Hello my Dear Friends,
I am having a problem with configuring my csv file. I am trying desperately
to configure the custom cdr format but in vain. I need some insight, if you
could help.
The cdr_custom.conf looks like this:-
Also, the cdr_custom.so is loaded.
;This is the format I am trying
;
; Mappings for custom config file
;
[mappings]
Master.csv =
${userid},${called_number},${CDR(uniqueid)},${CDR(answer)},${CDR(userfield)},${C
DR(duration)},
${SourceServer},${DestinationServer},${DestinationServerPort},${CDR(disposition)}
,${CDR(start)},
${CDR(billsec)}
Also, as mentioned in Readme.cdr shows that there is a module called cdr_csv2
and it allows us to create a custom cdr. I even tried this but it wouldnot work
as explained.( I had cdr_csv.so loaded for this but even this did not work ) .
Please correct me if I am going wrong somewhere.
I will be much obliged if you could help me.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Sweat saves blood, blood saves lives, and brains saves both.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Sweat saves blood, blood saves lives, and brains saves both.
___
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[Asterisk-Users] newbie cdr_custom and cdr_csv2 problem, please help
Hello my Dear Friends,
I am having a problem with configuring my csv file. I am trying desperately
to configure the custom cdr format but in vain. I need some insight, if you
could help.
The cdr_custom.conf looks like this:-
Also, the cdr_custom.so is loaded.
;This is the format I am trying
;
; Mappings for custom config file
;
[mappings]
Master.csv =
${userid},${called_number},${CDR(uniqueid)},${CDR(answer)},${CDR(userfield)},${CDR(duration)},
${SourceServer},${DestinationServer},${DestinationServerPort},${CDR(disposition)},${CDR(start)},
${CDR(billsec)}
Also, as mentioned in Readme.cdr shows that there is a module called cdr_csv2
and it allows us to create a custom cdr. I even tried this but it wouldnot work
as explained.( I had cdr_csv.so loaded for this but even this did not work ) .
Please correct me if I am going wrong somewhere.
I will be much obliged if you could help me.
With warm regards.
Vivek J. Joshi.
[EMAIL PROTECTED]
Trikon electronics Pvt. Ltd.
--Sweat saves blood, blood saves lives, and brains saves both.
___
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Re: [Asterisk-Users] Dial application newbie help
Dear Paul H., Thanks my dear friend, that worked. Thanks a lot for the help. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Sweat saves blood, blood saves lives, and brains saves both. ___ --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] Dial application newbie help
Hello friends, I am a newbie to asterisk , please help. I am receiving a phone from a sip server and I want to route it to another sip server. The problem is that the target sip server takes a # in the argument . I am trying to dial as exten = s,5,Dial([EMAIL PROTECTED]) dial the number at the ip address. When this gets executed, I get the following eror:- app_dial.c:698 dial_exec: Dial argument takes format (technology1/[device:]number1technology2/[device:]number2...|optional timeout) Please help me how do I resolve this problem. I have to send # or else my phone wont connect. Thanks for reading this. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Sweat saves blood, blood saves lives, and brains saves both. ___ --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] H323 compilation Help needed
Hi friend, You first need to have the correct verisons of pwlib and openh323 as mentioned in the readme file in ./channels/h323 directory. Note, they have to be the same versions, neither advanced nor otherwise, or else it wont compile. Then you ha ve to give a make from the channels/h323 directory and then give a make install from the asterisk base directory, i.e. /usr/src/asterisk or any other directory where you have untarred asterisk. I am sure of this because I have done it many times during our testing phases. But I recomend you that if you are using h323 protocol, then better use ooh323 which you can download from the internet. Use that package instead of h323 in asterisk. You will need different versions of pwlib and openh323 for it, but it works better than h323 in asterisk and oh323 in asterisk-addons. But remember, if at all you are using oh323 or ooh323, rename the conflicting verisons of chan_h323.so, or chan_oh323.so, or chan_ooh323.so from asterisk modules to something else. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Sweat saves blood, blood saves lives, and brains saves both. ___ --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] Newbie Problem With Agents
Hello Friends, I was trying to dial agents from a normal extension. My extensions.conf is configured as exten = 11,1,AgentCallbackLogin exten = 12,1,Dial(Agent/12) ;; configured in agents.conf as agent = 12,12, vivek exten = 13,1,Dial(SIP/13) ,, is configured in sip.conf When I dial 12 of call agents as exten = 2,1,Queue(sales_queue|t|||300) I get :- Jan 2 09:12:26 NOTICE[2782]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) But the agent is not busy. Kindly tell me where am I going wrong. Thank you all for reading this. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Optimism is a mania for saying things are well when one is in hell. ___ --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] Can we dial agents from extensions.conf
Hello friends, I wanted to ask if we can dial agents like the way we dial extensions. I wanted to try this because the users can login and others can dial them. If a person has not logged in, he isnt avalaible. I dont want to put people in a queue. Has anyone tried this before? I was trying to do it but was unsuccessful. Please tell me if there is a tweak or a workaround for this. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Optimism is a mania for saying things are well when one is in hell. ` ___ --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] Can we dial agents from extensions.conf
Thanks a lot Mr. Alexander Lopez for your prompt attension. I tried the same thing but it wouldnot happen. I use it as:- exten = 12,1,Dial(Agent/12) exten = 12,2,Hangup where agent 12 is configured as :- agent = 12,12, vivek After the agent is logged in on extension no12 as follows Callback Agent '12' logged in on 12 I try to dial 12 from another sip phone and get this:- -- Executing Dial(SIP/62-c24e, Agent/12) in new stack -- outgoing agentcall, to agent '12', on 'Local/[EMAIL PROTECTED],1' -- Called 12 -- Executing Dial(Local/[EMAIL PROTECTED],2, Agent/12) in new stack Dec 30 14:26:54 NOTICE[13289]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) == Everyone is busy/congested at this time (1:1/0/0) -- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack == Spawn extension (default, 12, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' -- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack == Spawn extension (default, h, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' == No one is available to answer at this time (1:0/0/0) -- Executing Hangup(SIP/62-c24e, ) in new stack == Spawn extension (inoffice, 12, 2) exited non-zero on 'SIP/62-c24e' -- Executing Hangup(SIP/62-c24e, ) in new stack == Spawn extension (inoffice, h, 1) exited non-zero on 'SIP/62-c24e' I am unable to figure out why it is happening like this. They are all in the same context. Also, the agent is not busy. Also, I wonder why it says Unable to creat0e chanel of type 'Agent' cause user busy. Do you have any idea why is it happening so? I tried to tweak in but was not successful. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Optimism is a mania for saying things are well when one is in hell. Alexander Lopez wrote: There are options for queues.conf to not allow callers to join a queue if no members are logged in, also you can 'call an agent' with the agent channel, (IE agent/100) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 30, 2005 7:17 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can we dial agents from extensions.conf Hello friends, I wanted to ask if we can dial agents like the way we dial extensions. I wanted to try this because the users can login and others can dial them. If a person has not logged in, he isnt avalaible. I dont want to put people in a queue. Has anyone tried this before? I was trying to do it but was unsuccessful. Please tell me if there is a tweak or a workaround for this. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Optimism is a mania for saying things are well when one is in hell. ` ___ --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] Can we dial agents from extensions.conf
Hello Mr. Lopez, I am using 5 Welltech fxs boxes which are configured on SIP. These are connected in lan with my asterisk server. I have configured these boxes with sip as follows:= [62] type=friend username=62 secret=1234 context=inoffice host=dynamic dtmfmode=info canreinvite=no callgroup=1 pickupgroup=1 All others are configured in a similar fashion. I am able to dial to a particular SIP phone using exten = 30,1,Dial(SIP/62) ; 62 is the username above. The extensions.conf registers users as follows:- exten = 111,1,AgentCallBackLogin The person goes on any telephone instrument connected to one of these boxes and dials 111. He then enters the digits as:- User=12 Password=12 Extension=12 And the agent logs in on a SIP phone. The basic idea is that I want everyone to have an extension and to sit wherever they like. Irrespective of their location, they can be reached by their extension. Also, I want a call centre type calling system and thats why I am using agents. In extensions.conf, the extension 12 was tried as follows:- exten = 12,1,Dial(Agent/12) and then I got the error message I posted. In agents.conf, I have configured the agent 12 as follows:- agent = 12,12,vivek I am not able to figure out why would not it dial agent 12. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Optimism is a mania for saying things are well when one is in hell. Alexander Lopez wrote: Can you tell me how agent 12 is logging in, Zap, Iax, SIP??? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 30, 2005 9:35 AM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Can we dial agents from extensions.conf Thanks a lot Mr. Alexander Lopez for your prompt attension. I tried the same thing but it wouldnot happen. I use it as:- exten = 12,1,Dial(Agent/12) exten = 12,2,Hangup where agent 12 is configured as :- agent = 12,12, vivek After the agent is logged in on extension no12 as follows Callback Agent '12' logged in on 12 I try to dial 12 from another sip phone and get this:- -- Executing Dial(SIP/62-c24e, Agent/12) in new stack -- outgoing agentcall, to agent '12', on 'Local/[EMAIL PROTECTED],1' -- Called 12 -- Executing Dial(Local/[EMAIL PROTECTED],2, Agent/12) in new stack Dec 30 14:26:54 NOTICE[13289]: app_dial.c:1011 dial_exec_full: Unable to create channel of type 'Agent' (cause 17 - User busy) == Everyone is busy/congested at this time (1:1/0/0) -- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack == Spawn extension (default, 12, 2) exited non-zero on 'Local/[EMAIL PROTECTED],2' -- Executing Hangup(Local/[EMAIL PROTECTED],2, ) in new stack == Spawn extension (default, h, 1) exited non-zero on 'Local/[EMAIL PROTECTED],2' == No one is available to answer at this time (1:0/0/0) -- Executing Hangup(SIP/62-c24e, ) in new stack == Spawn extension (inoffice, 12, 2) exited non-zero on 'SIP/62-c24e' -- Executing Hangup(SIP/62-c24e, ) in new stack == Spawn extension (inoffice, h, 1) exited non-zero on 'SIP/62-c24e' I am unable to figure out why it is happening like this. They are all in the same context. Also, the agent is not busy. Also, I wonder why it says Unable to creat0e chanel of type 'Agent' cause user busy. Do you have any idea why is it happening so? I tried to tweak in but was not successful. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Optimism is a mania for saying things are well when one is in hell. Alexander Lopez wrote: There are options for queues.conf to not allow callers to join a queue if no members are logged in, also you can 'call an agent' with the agent channel, (IE agent/100) -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED] Sent: Friday, December 30, 2005 7:17 AM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] Can we dial agents from extensions.conf Hello friends, I wanted to ask if we can dial agents like the way we dial extensions. I wanted to try this because the users can login and others can dial them. If a person has not logged in, he isnt avalaible. I dont want to put people in a queue. Has anyone tried this before? I was trying to do it but was unsuccessful. Please tell me if there is a tweak or a workaround for this. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Optimism is a mania for saying things are well when one is in hell. ` ___ --Bandwidth and Colocation provided by Easynews.com
[Asterisk-Users] Re: Newbie question
Thanks Mr.Miano Thanks a lot. Now I think I wont have to bother about balming all my problems to zapata. I have also succeeded quite a bit and installed a basic PBX system without it. Thanks a lot again. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Optimism is a mania for saying things are well when one is in hell. Giovanni Miano wrote: I dont need to configure zaptel device, you dont use it :) 2005/11/30, [EMAIL PROTECTED] [EMAIL PROTECTED]: Hello friends, I am using asterisk 1.2 with ooh323. I am using sip and h323 phones. My question is I am using a Welltech FXO box and ip phones by Welltech. Do I still need to configure zapata.conf and zaptel.conf which I read in the documentation from asterisk pdf file downoladed from asterisk.org ? I think I dont because I dont use a digium card but do I have to still confugure for FXO and FXS ports? Kindly help me solving my doubt. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Truth springs from argument amongst friends. ___ --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 -- Giovanni Miano ___ --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] Newbie question
Hello friends, I am using asterisk 1.2 with ooh323. I am using sip and h323 phones. My question is I am using a Welltech FXO box and ip phones by Welltech. Do I still need to configure zapata.conf and zaptel.conf which I read in the documentation from asterisk pdf file downoladed from asterisk.org ? I think I dont because I dont use a digium card but do I have to still confugure for FXO and FXS ports? Kindly help me solving my doubt. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Truth springs from argument amongst friends. ___ --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: think people dont help that easily
With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Truth springs from argument amongst friends. [EMAIL PROTECTED] wrote: Hello friends, I have a strange problem. I am using asterisk 1.2 and asterisk addons 1.2. I have three SIP phones and one H323 phones connected to asterisk. The problem is that when I dial an invalid extension from H323 phones, I get the invalid extension message with exten = i... in that context but this does not happen with the SIP phones. All I get is something like an engaged tone from the SIP phones. Also I am able to dial and transfer between SIP and H323 phones. I am not able to figure out whats wrong. None of them are behind the NAT. All of them and the asterisk server are on private-ip. I also tried sip debug from the command line and dial an invlaid extension from the SIP phone and get nothing but a SIP/2.0 404 Not Found in the o/p. But it then dosent fall to the exten = i or exten = s. My conf. files are as under:- extensions.conf:- [incoming] exten = s,1,Answer ; Answer the line. exten = s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds. exten = s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds. exten = s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message. exten = s,n,WaitExten(5) ; Wait for an extension to be dialed. exten = s,n,Dial(SIP/192.168.1.196,100,t) , Dial the operator. exten = i,1,Playback(invalid) ; That's not valid, try again. [default] include = incoming ; Instead of demo in the sample, there is incoming. [testing] include = parkedcalls exten = s,1,Playback(invalid) ; When this is present, invalid extension from h323 comes here or ;;; exten = i,1,Playback(invalid) ;;;even this did not work. ;; H323 Phones ;; exten = 61,1,Dial(OOH323/192.168.1.194,20|t) ;ip=h323 ;; SIP Phones ;; exten = 62,1,Dial(SIP/62,20|t);new-gray=sip exten = 63,1,Dial(SIP/63,20|t);old-gray=sip exten = 64,1,Dial(SIP/64,20|t);ip=sip ooh323.conf:- context=testing disallow=all allow=ulaw allow=alaw dtmfmode=h245alphanumeric [61] type=friend ip=192.168.1.194 context=testing sip.conf:- [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=alaw allow=ulaw musicclass=default dtmfmode = rfc2833 [63] type=friend context=testing ; context above where the extensions dialable by this are defined. username=63 secret=1234 host=dynamic defaultip=192.168.1.192 ; ip address of this phone canreinvite=no callgroup=1 ; We are in caller groups 1 pickupgroup=1 ; We can do call pick-p for call group 1 ;; rest of the sip users are configured in the same way. Help will be very much appreciated. Kindly help. I am totally confused as to where the fault is. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Truth springs from argument amongst friends. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Sip dosenot fall to default 's' , STRANGE?
Hello friends, I have a strange problem. I am using asterisk 1.2 and asterisk addons 1.2. I have three SIP phones and one H323 phones connected to asterisk. The problem is that when I dial an invalid extension from H323 phones, I get the invalid extension message with exten = i... in that context but this does not happen with the SIP phones. All I get is something like an engaged tone from the SIP phones. Also I am able to dial and transfer between SIP and H323 phones. I am not able to figure out whats wrong. None of them are behind the NAT. All of them and the asterisk server are on private-ip. I also tried sip debug from the command line and dial an invlaid extension from the SIP phone and get nothing but a SIP/2.0 404 Not Found in the o/p. But it then dosent fall to the exten = i or exten = s. My conf. files are as under:- extensions.conf:- [incoming] exten = s,1,Answer ; Answer the line. exten = s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds. exten = s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds. exten = s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message. exten = s,n,WaitExten(5) ; Wait for an extension to be dialed. exten = s,n,Dial(SIP/192.168.1.196,100,t) , Dial the operator. exten = i,1,Playback(invalid) ; That's not valid, try again. [default] include = incoming ; Instead of demo in the sample, there is incoming. [testing] include = parkedcalls exten = s,1,Playback(invalid) ; When this is present, invalid extension from h323 comes here or ;;; exten = i,1,Playback(invalid) ;;;even this did not work. ;; H323 Phones ;; exten = 61,1,Dial(OOH323/192.168.1.194,20|t) ;ip=h323 ;; SIP Phones ;; exten = 62,1,Dial(SIP/62,20|t);new-gray=sip exten = 63,1,Dial(SIP/63,20|t);old-gray=sip exten = 64,1,Dial(SIP/64,20|t);ip=sip ooh323.conf:- context=testing disallow=all allow=ulaw allow=alaw dtmfmode=h245alphanumeric [61] type=friend ip=192.168.1.194 context=testing sip.conf:- [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all allow=alaw allow=ulaw musicclass=default dtmfmode = rfc2833 [63] type=friend context=testing ; context above where the extensions dialable by this are defined. username=63 secret=1234 host=dynamic defaultip=192.168.1.192 ; ip address of this phone canreinvite=no callgroup=1 ; We are in caller groups 1 pickupgroup=1 ; We can do call pick-p for call group 1 ;; rest of the sip users are configured in the same way. Help will be very much appreciated. Kindly help. I am totally confused as to where the fault is. With warm regards. Vivek J. Joshi. [EMAIL PROTECTED] Trikon electronics Pvt. Ltd. --Truth springs from argument amongst friends. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
