Re: [asterisk-users] Push presence from one asterisk to another
Thank you for the information. I think the Junghanns esel stuff is what I would need right now. I read that dundi could be used later for distributed presence. Are those plans still valid? Loic. On Fri, 2008-09-26 at 17:31 -0700, Kevin P. Fleming wrote: Philipp Kempgen wrote: Junghanns' BriStuff can do it via ESEL (extension state export logic). Basically that's a connection between the AMIs. In Asterisk 1.6 you could do it via DEVSTATE(). http://www.asterisk.org/blog/8 Asterisk 1.6.1 will have distributed device state as well, although the current mechanisms for distribution (OpenAIS) are designed only for use over a low latency LAN connection, not VPNs or WAN links. ___ -- 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
Re: [asterisk-users] Dial issue
Hi, can you please confirm that DTMF is working properly or not? Thanks, Max Alex Voip Developer On Sat, Sep 27, 2008 at 12:24 AM, equis software [EMAIL PROTECTED]wrote: Hi, when I make a call I need that the caller can** hang up by dialing ***(H option in Dial command), the call but it don´t work. Command EXEC DIAL Zap/g1/433391|20|H In CLI... -- AGI Script Executing Application: (DIAL) Options: (Zap/g1/433391|20|H) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/433391 -- Zap/1-1 is ringing -- Zap/1-1 answered SIP/510093-082160f0 (--- At this moment I press * several times, but nothing happens Then I hung up the phone--) -- Hungup 'Zap/1-1' Any Ideas? ___ -- 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 ___ -- 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
Re: [asterisk-users] Extremely OT: I need someone who can parse a MS Word or PDF or RTF document
On Fri, Sep 26, 2008 at 10:59 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: randulo schrieb: take it on. Basically this is taking a human readable text and turning it into a bunch of database SQL inserts. Out of curiosity: Why? Indeed, I could probably hire people in the third world (or maybe soon in the US!!!) to read the doc and type it into a database, but heck, that'd put programmers out of business. I guess the database having 40,000 searchable notes is the real answer though. Awful hard to wade through a bunch word docs using text search. I have a couple of promising replies. Thanks for your patience, folks, any further comments please make them off list. /r ___ -- 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] test call generator
Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam ___ -- 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
Re: [asterisk-users] test call generator
Are you looking for inbound or outbound. I can get you free inbound test DID. LMK Jai www.didforesale.com On 9/27/08, Sam Tam [EMAIL PROTECTED] wrote: Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam -- Sent from Gmail for mobile | mobile.google.com ___ -- 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
Re: [asterisk-users] test call generator
Sam Tam wrote: Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam ___ -- 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 Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com ___ -- 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] Problem with pickup extension *8 from features.conf using IAX
Hello list I am trying to configure a PBX using Asterisk. The problem I am havong is the following: I want to use the *8 from features.conf to pickup any ringing extension from a group, becouse I want to put the users in call queues and I want anybody from the company to be able to pick a ringing channel, even if is in a queue. Whwn using Sip protocol for the users, everithing is going fine, I can pickup any ringing extension from the group using *8. But the problem appears when I am using IAX protocol. When issuing *8 from the IAX phone, asterisk tryes to find the *8 in the dialling rules returning: *CLI -- Registered IAX2 '40' (AUTHENTICATED) at 10.0.0.30:4569 [Sep 27 12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process: Rejected connect attempt from 10.0.0.30, request '[EMAIL PROTECTED]' does not exist This I think is wrong, is something like asterisk cannot read from features. With the same setting, when using SIP, i get: *CLI == Using SIP RTP CoS mark 5 [Sep 27 12:06:23] NOTICE[19802]: chan_sip.c:17092 handle_request_invite: Nothing to pick up for [EMAIL PROTECTED] and it works ok. I am wondering if any had this problem before and if you can help me figure it out(how to make it work--or if is a bug), or find a sollution using the app pickup. I tryed using asterisk 1.4.13, asterisk 1.4.21.2, asterisk 1.6-rc6 and always the same problem ocurs. Regards, Cosmin ___ -- 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
Re: [asterisk-users] Extremely OT: I need someone who can parse a MS Word or PDF or RTF document
On Sat, Sep 27, 2008 at 3:41 AM, randulo [EMAIL PROTECTED] wrote: On Fri, Sep 26, 2008 at 10:59 PM, Philipp Kempgen [EMAIL PROTECTED] wrote: randulo schrieb: take it on. Basically this is taking a human readable text and turning it into a bunch of database SQL inserts. Out of curiosity: Why? Indeed, I could probably hire people in the third world (or maybe soon in the US!!!) to read the doc and type it into a database, but heck, that'd put programmers out of business. I guess the database having 40,000 searchable notes is the real answer though. Awful hard to wade through a bunch word docs using text search. I have a couple of promising replies. Thanks for your patience, folks, any further comments please make them off list. /r Have you tried any of the many Freelancers sites before going for the OT post to this list? What is your budget? Thanks, Steve Totaro ___ -- 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
Re: [asterisk-users] test call generator
On Sat, Sep 27, 2008 at 4:38 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Sam Tam wrote: Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com It's not free but if you want some good ideas for features, or don't mind paying, there is the Empirix Hammer. http://www.empirix.com/ Thanks, Steve Totaro ___ -- 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] running out of disk space
I am a very junior Asterisk user, and it has been some time since I have used linuxNow that I got that out of my way I was curious if someone could point me in the right direction to find out how to clean up my asterisk server. The hard drive has 150 gigs on it and it only has 200 mb free. I am cleaning out the /var/logs just to keep the system running, but I need to find out if asterisk is dumping voice messages somewhere that should have been deleted, or some other issue that might be causing this space to be used up??? Any thoughts??? ___ -- 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
Re: [asterisk-users] test call generator
What you are looking for is SIPP: http://sipp.sourceforge.net/ It won't intrinsically tell you anything about the data; it's up to you to appropriate the findings. But it accomplishes the generation of traffic (and dummy media!) on a technical level. Igor Hernandez wrote: Sam Tam wrote: Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam ___ -- 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 Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: (+1) (678) 954-0670 Direct : (+1) (678) 954-0671 Mobile : (+1) (706) 338-8599 ___ -- 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
Re: [asterisk-users] Dial issue
DTMF is working properly because in other part of my eagi script I use this DTMF. Thanks Doverli On Sat, Sep 27, 2008 at 3:16 AM, Max Alex [EMAIL PROTECTED] wrote: Hi, can you please confirm that DTMF is working properly or not? Thanks, Max Alex Voip Developer On Sat, Sep 27, 2008 at 12:24 AM, equis software [EMAIL PROTECTED]wrote: Hi, when I make a call I need that the caller can** hang up by dialing ** * (H option in Dial command), the call but it don´t work. Command EXEC DIAL Zap/g1/433391|20|H In CLI... -- AGI Script Executing Application: (DIAL) Options: (Zap/g1/433391|20|H) -- Requested transfer capability: 0x00 - SPEECH -- Called g1/433391 -- Zap/1-1 is ringing -- Zap/1-1 answered SIP/510093-082160f0 (--- At this moment I press * several times, but nothing happens Then I hung up the phone--) -- Hungup 'Zap/1-1' Any Ideas? ___ -- 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 ___ -- 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 ___ -- 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
Re: [asterisk-users] running out of disk space
You could use a find command and search for large files but that won’t help if there are many small files in a directory. You can use du and pipe it into sort -n du | sort -n | tail -1000 | more that will give you the 1000 LARGEST directories. You can go from there Alex Kindly consider the environment before printing this e-mail. -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mr surfit Sent: Saturday, September 27, 2008 6:58 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] running out of disk space I am a very junior Asterisk user, and it has been some time since I have used linuxNow that I got that out of my way I was curious if someone could point me in the right direction to find out how to clean up my asterisk server. The hard drive has 150 gigs on it and it only has 200 mb free. I am cleaning out the /var/logs just to keep the system running, but I need to find out if asterisk is dumping voice messages somewhere that should have been deleted, or some other issue that might be causing this space to be used up??? Any thoughts??? ___ -- 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 ___ -- 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] Philippines
Anyone on the list involved in installing Asterisk in the Philippines - preferably someone with SugarCRM integration experience? I have a friend who is setting up a domestic outbound call center with about 20 agents initially looking for a simple low cost implementation. Email me with reference information and I'll send you contact details. Regards, Dean Collins [EMAIL PROTECTED] +1-212-203-4357 (New York) +61-2-9016-5642 (Sydney) http://www.Cognation.net http://www.Cognation.net/profile ___ -- 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] Asterisk and VoIP educational resources
Hi, Next week on the VUC, we will be reviewing as many ways to bullet proof your entry into the mailing list or the IRC channel. We all know RTFM is the first step. It is in the interest of everyone in the community to make as many resources know as possible. If you have a site or book that is not on the list below, please either post to this thread or send it to me using http://delicious.com with the tag for:voipusersconference http://voip-info.org http://lists.digium.com http://www.asteriskblog.com/ http://www.asteriskdocs.org/ http://www.voipusersconference.org http://www.disruptivetelephony.com/ http://www.mgraves.org/voip/ http://www.voip-news.com http://www.the-asterisk-book.com http://www.voipspeak.net/ http://www.venturevoip.com/news.php http://www.asteriskguru.com/ http://asterisk.net.au/ Note that nearly every blog starts with Asterisk + Skype news :) Besides the above and many more we'll talk about, there are countless niche resources like Lumenvox and their tutorials in speech recognition Please add your resources to the list and better yet, send them via delicious.com /r ps. I wish I had a recording of my entry into the world of asterisk via #asterisk. The words crack pipe and moose .p... were an integral part. Ring any bells? As I've said many times, my favorite intro is still John Todd's two articles: http://www.onlamp.com/pub/a/onlamp/2003/07/03/asterisk.html http://www.onlamp.com/pub/a/onlamp/2004/01/22/asterisk2.html They are dated, yes, but there's more info per inch there than a lot of other sites! Thanks, John! Another great (but again outdated) text: http://automated.it/guidetoasterisk.htm Someone needs to start these initiatives again for 1.6. ___ -- 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
Re: [asterisk-users] test call generator
You actually using that steve? Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, September 27, 2008 6:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] test call generator On Sat, Sep 27, 2008 at 4:38 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Sam Tam wrote: Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com It's not free but if you want some good ideas for features, or don't mind paying, there is the Empirix Hammer. http://www.empirix.com/ Thanks, Steve Totaro ___ -- 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
Re: [asterisk-users] test call generator
Unforunately it is outbound -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jai Rangi Sent: Saturday, September 27, 2008 4:05 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] test call generator Are you looking for inbound or outbound. I can get you free inbound test DID. LMK Jai www.didforesale.com On 9/27/08, Sam Tam [EMAIL PROTECTED] wrote: Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam -- Sent from Gmail for mobile | mobile.google.com ___ -- 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 ___ -- 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
Re: [asterisk-users] Problem with pickup extension *8 from features.conf using IAX
already tested with an exten?
ex:
exten = _*8.,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _*8.,n,Hangup()
2008/9/27 coco [EMAIL PROTECTED]
Hello list
I am trying to configure a PBX using Asterisk.
The problem I am havong is the following: I want to use the *8 from
features.conf to pickup any ringing extension from a group, becouse I want
to put the users in call queues and I want anybody from the company to be
able to pick a ringing channel, even if is in a queue.
Whwn using Sip protocol for the users, everithing is going fine, I can
pickup any ringing extension from the group using *8.
But the problem appears when I am using IAX protocol. When issuing *8 from
the IAX phone, asterisk tryes to find the *8 in the dialling rules
returning:
*CLI -- Registered IAX2 '40' (AUTHENTICATED) at 10.0.0.30:4569
[Sep 27 12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process: Rejected
connect attempt from 10.0.0.30, request '[EMAIL PROTECTED]' [EMAIL
PROTECTED]does not exist
This I think is wrong, is something like asterisk cannot read from
features.
With the same setting, when using SIP, i get:
*CLI == Using SIP RTP CoS mark 5
[Sep 27 12:06:23] NOTICE[19802]: chan_sip.c:17092 handle_request_invite:
Nothing to pick up for [EMAIL PROTECTED]
and it works ok.
I am wondering if any had this problem before and if you can help me figure
it out(how to make it work--or if is a bug), or find a sollution using the
app pickup.
I tryed using asterisk 1.4.13, asterisk 1.4.21.2, asterisk 1.6-rc6 and
always the same problem ocurs.
Regards,
Cosmin
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shazaum.googlepages.com
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Re: [asterisk-users] Set A-Number in Sip Header
I'm not sure if this is what you want, but set(CALLERID(num)=) 2008/9/27 Michael Litzel [EMAIL PROTECTED] Hi, ours Alcatel is directly with asterisk connected via pmx. Ours sip carrier needs our ISDN PMX head number in the Sip header. The CALLERID (num) should be set however on anonymous, which is not indicated at the B-connection a number. Is there a possibility of the Sip here header manipulation in extension.conf? Asterisk Version 1.4 Thanks for answer, Michael Litzel __ Hinweis von ESET NOD32 Antivirus, Signaturdatenbank-Version 3475 (20080926) __ E-Mail wurde geprüft mit ESET NOD32 Antivirus. http://www.eset.com __ Hinweis von ESET NOD32 Antivirus, Signaturdatenbank-Version 3475 (20080926) __ E-Mail wurde geprüft mit ESET NOD32 Antivirus. http://www.eset.com ___ -- 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 user number: 1099 Linux user: #443184 shazaum.googlepages.com ___ -- 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
Re: [asterisk-users] Set A-Number in Sip Header
but this does not solve? set(CALLERID(num)=anonymous) 2008/9/27 Michael Litzel [EMAIL PROTECTED] Hi Shazaum, Thanks for the answer, but that is not which I means, ours Sip Peer is an OpenSer. We must manipulate the Sip Header, the Source Number. The CallerID must be anonymous. Michael *Von:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *Im Auftrag von *Shazaum *Gesendet:* Samstag, 27. September 2008 19:19 *An:* Asterisk Users Mailing List - Non-Commercial Discussion *Betreff:* Re: [asterisk-users] Set A-Number in Sip Header I'm not sure if this is what you want, but set(CALLERID(num)=) 2008/9/27 Michael Litzel [EMAIL PROTECTED] Hi, ours Alcatel is directly with asterisk connected via pmx. Ours sip carrier needs our ISDN PMX head number in the Sip header. The CALLERID (num) should be set however on anonymous, which is not indicated at the B-connection a number. Is there a possibility of the Sip here header manipulation in extension.conf? Asterisk Version 1.4 Thanks for answer, Michael Litzel __ Hinweis von ESET NOD32 Antivirus, Signaturdatenbank-Version 3475 (20080926) __ E-Mail wurde geprüft mit ESET NOD32 Antivirus. http://www.eset.com __ Hinweis von ESET NOD32 Antivirus, Signaturdatenbank-Version 3475 (20080926) __ E-Mail wurde geprüft mit ESET NOD32 Antivirus. http://www.eset.com ___ -- 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 user number: 1099 Linux user: #443184 shazaum.googlepages.com __ Hinweis von ESET NOD32 Antivirus, Signaturdatenbank-Version 3476 (20080927) __ E-Mail wurde geprüft mit ESET NOD32 Antivirus. http://www.eset.com ___ -- 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 user number: 1099 Linux user: #443184 shazaum.googlepages.com ___ -- 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
Re: [asterisk-users] test call generator
There is no reason that outbound cannot also be inbound.. mind wandering to the mobius strip I am not using it but I do have plans to shortly. I think if you want any kind of real testing and validation, then a product like this is almost required. As Alex noted, you could use SIPp, you could also use originate, .call files, and other methods, but do you get anything useful except some info from top and maybe a self monitored call or two? Thanks, Steve Totaro On Sat, Sep 27, 2008 at 12:39 PM, Sam Tam [EMAIL PROTECTED] wrote: You actually using that steve? Sam -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro Sent: Saturday, September 27, 2008 6:26 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] test call generator On Sat, Sep 27, 2008 at 4:38 AM, Igor Hernandez [EMAIL PROTECTED] wrote: Sam Tam wrote: Hello everyone I am trying to look for a free test call generator that will get me some stats like PDD, ASR and call quality etc on each route. As well as do test at every interval too If you know something like this please enlighten me. Sam Hey Sam, I've been looking for such a tool also. I can't seem to find a tool that does those things. If nothing comes up in the next couple of weeks I'm going to code something up, I wouldn't mind letting you and anyone else who might be interested have the source once its done. Let me know if you find anything thats already out there in the meantime, might just save me a few hours of work. Regards, -- Igor Hernandez Escape Communications http://www.escapetel.com It's not free but if you want some good ideas for features, or don't mind paying, there is the Empirix Hammer. http://www.empirix.com/ Thanks, Steve Totaro ___ -- 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 ___ -- 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
Re: [asterisk-users] running out of disk space
Probably its saving its calls in wav format Just check your recording directory , probably a lot of wav files in there. rm *.wav -f will then do the trick -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mr surfit Sent: Samstag, 27. September 2008 12:58 To: asterisk-users@lists.digium.com Subject: [asterisk-users] running out of disk space I am a very junior Asterisk user, and it has been some time since I have used linuxNow that I got that out of my way I was curious if someone could point me in the right direction to find out how to clean up my asterisk server. The hard drive has 150 gigs on it and it only has 200 mb free. I am cleaning out the /var/logs just to keep the system running, but I need to find out if asterisk is dumping voice messages somewhere that should have been deleted, or some other issue that might be causing this space to be used up??? Any thoughts??? ___ -- 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 No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.169 / Virus Database: 270.7.3/1694 - Release Date: 26.09.2008 18:55 ___ -- 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
Re: [asterisk-users] iPhone Sip App
There is allready a SIPGATE client. Closed for use with sipgate only, but there will be more shortly... From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Forrest Beck Sent: Samstag, 27. September 2008 04:11 To: Asterisk Users List Subject: [asterisk-users] iPhone Sip App Has anyone seen or know of a iphone/ipod sip client that may be in the works? No virus found in this incoming message. Checked by AVG - http://www.avg.com Version: 8.0.169 / Virus Database: 270.7.2/1690 - Release Date: 25.09.2008 19:23 ___ -- 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] FW: Google Alert - dean collins
Who is Chris Langford in Huntsville Alabama and is he seeking Digium's permission in order to report the asterisk mailing lists out onto the internet http://asteriskbizrss.blogspot.com/ http://www.blogger.com/profile/04174728129647374395 What can be done to stop people doing this and making money out of selling ads on these crappy blogsites? Cheers, Dean From: Google Alerts [mailto:[EMAIL PROTECTED] Sent: Saturday, 27 September 2008 2:38 PM To: Dean Collins Subject: Google Alert - dean collins Google Blogs Alert for: dean collins [asterisk-biz] Philippines http://asteriskbizrss.blogspot.com/2008/09/asterisk-biz-philippines.htm l By Chris Langford(Chris Langford) I have a friend who is setting up a domestic outbound call center with about 20 agents initially looking for a simple low cost implementation. Email me with reference information and I'll send you contact details. Regards,. Dean Collins ... Asterisk Biz - http://asteriskbizrss.blogspot.com/ http://asteriskbizrss.blogspot.com/ This as-it-happens Google Alert is brought to you by Google. Remove http://www.google.com/alerts/remove?s=EAWStCkc7lwdmf82QEbYdZMhl=en gl= this alert. Create http://www.google.com/alerts?hl=engl= another alert. Manage http://www.google.com/alerts/manage?hl=engl= your alerts. ___ -- 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
Re: [asterisk-users] FW: Google Alert - dean collins
On Sat, 2008-09-27 at 15:11 -0400, Dean Collins wrote: Who is Chris Langford in Huntsville Alabama and is he seeking Digium’s permission in order to report the asterisk mailing lists out onto the internet Digium's permission? Does Digium own the copyright on what I write? I think not. I do. I don't recall assigning any copyright when I joined the list. On the other hand I think there is an implicit assignment to the public domain when you contribute to a ML. http://asteriskbizrss.blogspot.com/ What can be done to stop people doing this and making money out of selling ads on these crappy blogsites? What ads? I don't see any ads. Now if you really want to complain about this particular subject you could start with osdir.com: http://osdir.com/ml/isdn.i4l.user/2005-09/msg7.html Although I'm really not sure what you are in a tizzy about. It seems a fair trade to get some ad revenue to support the cost of the archiving which is a convenience to the people who don't want to go to the expense of archiving everything they might be interested in ever reading themselves. I find it surprising that the US economy is on the brink of depression with an unprecedented $700 billion (and that _billion_ with a B) bailout of the credit economy and you are complaining about somebody possibly recouping his expenses for what is arguably a public service. ~sigh~ b. signature.asc Description: This is a digitally signed message part ___ -- 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
Re: [asterisk-users] Asterisk 1.4 or 1.6
Yes, that is what I did, I used overlay but I had a hard time to unmak it. The 1.4 is not even in the portage unstable and it was masked in: /usr/portage/profile/package.mask True. The asterisk packages in the voip overlay aren't particularly up-to-date. If I had more knowledge of Portage and how it works, I'd help out, but alas, I don't, and don't really have time to learn. For some of the packages you need to edit the ebuild file and add ~amd64 ~x86 (depending on your architecture) to the ARCH= line, then rebuild the digests on the ebuilds. Regards, Chris ___ -- 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
Re: [asterisk-users] FW: Google Alert - dean collins
Dean, You retain copyright to your e-mail. Send a DMCA take down notice to blogspot. The real problem is the ad networks, no ads, no incentive to profit from stealing content for ad revenue. It's funny how some advertisers with really good search engines can't seem to use the same search engine to figure when they are profiting from stolen content. On Sep 27, 2008, at 12:11 PM, Dean Collins wrote: Who is Chris Langford in Huntsville Alabama and is he seeking Digium’s permission in order to report the asterisk mailing lists out onto the internet http://asteriskbizrss.blogspot.com/ http://www.blogger.com/profile/04174728129647374395 What can be done to stop people doing this and making money out of selling ads on these crappy blogsites? Cheers, Dean From: Google Alerts [mailto:[EMAIL PROTECTED] Sent: Saturday, 27 September 2008 2:38 PM To: Dean Collins Subject: Google Alert - dean collins Google Blogs Alert for: dean collins [asterisk-biz] Philippines By Chris Langford(Chris Langford) I have a friend who is setting up a domestic outbound call center with about 20 agents initially looking for a simple low cost implementation. Email me with reference information and I’ll send you contact details. Regards,. Dean Collins ... Asterisk Biz - http://asteriskbizrss.blogspot.com/ This as-it-happens Google Alert is brought to you by Google. Remove this alert. Create another alert. Manage your alerts. ___ -- 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 -- Eric Chamberlain, Founder RF.com - http://RF.com/ ___ -- 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
Re: [asterisk-users] FW: Google Alert - dean collins
On Sat, Sep 27, 2008 at 03:11:10PM -0400, Dean Collins wrote: Who is Chris Langford in Huntsville Alabama and is he seeking Digium's permission in order to report the asterisk mailing lists out onto the internet http://asteriskbizrss.blogspot.com/ http://www.blogger.com/profile/04174728129647374395 What can be done to stop people doing this and making money out of selling ads on these crappy blogsites? H How did you get this information? Isn't it because someone else indexed content they did not generate? That someone even makes money from selling ads for search results! -- 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 -- 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] rtpkeepalive problem ?
Hello, I'm having a problem when registering a x-lite to my asterisk server and bridging the xlite SIP channel to a PSTN SIP channel; in such case, the audio paths are only created x seconds after rtpkeepalive expires. If I set rtpkeepalive to 0, I never get the audio paths. I wiresharked it and can see the ComfortNoise packet beeing sent 10 seconds later (my rtpkeepalive value); I tried using nat=yes, withtout any change. Any help on where I should start? Thanks. Sebastien. ___ -- 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
Re: [asterisk-users] FW: Google Alert - dean collins
On Sat, Sep 27, 2008 at 3:46 PM, Tzafrir Cohen [EMAIL PROTECTED]wrote: On Sat, Sep 27, 2008 at 03:11:10PM -0400, Dean Collins wrote: Who is Chris Langford in Huntsville Alabama and is he seeking Digium's permission in order to report the asterisk mailing lists out onto the internet http://asteriskbizrss.blogspot.com/ http://www.blogger.com/profile/04174728129647374395 What can be done to stop people doing this and making money out of selling ads on these crappy blogsites? H How did you get this information? Isn't it because someone else indexed content they did not generate? That someone even makes money from selling ads for search results! -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED][EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir Dean is just mad that he is not the one with the money making scheme this time. While copyright law makes it technically illegal to reproduce almost any new creative work (other than under fair use) without permission, if the work is unregistered and has no real commercial value, it gets very little protection. The author in this case can sue for an injunction against the publication, *actual* damages from a violation, and possibly court costs. Actual damages means actual money potentially lost by the author due to publication, plus any money gained by the defendant. But if a work has no commercial value, such as a typical E-mail message or conversational USENET posting, the actual damages will be zero. Only the most vindictive (and rich) author would sue when no damages are possible, and the courts don't look kindly on vindictive plaintiffs, unless the defendants are even more vindictive. http://www.templetons.com/brad/copymyths.html Thanks, Steve Totaro ___ -- 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
Re: [asterisk-users] Bizarre international call problem.
You have handsets connected to your proprietary PBX. Most domestic things you dial on your proprietary PBX handsets get passed directly through to your asterisk box without getting mangled by your proprietary PBX. International calls that are prefixed by 011 are getting mangled by your proprietary PBX. Are you already getting to what I'm going to suggest? Modify your proprietary PBX to not mangle your international calls. Well, I really like that idea, but there's one small problem: outbound calls work just fine when the Asterisk system is removed from the equation. I'm now leaning slightly toward there being T1 funkiness between the PoS and the Asterisk box... but without a T1 protocol analyzer, it's kinda hard to be sure. Hopefully, I can get a friend over (with his) to help out. I guess -- maybe -- they could be playing games, and having the PSTN assume that calls sent out over channel X are international, but that's now sounding super-duper improbable. Thanks... -Ken ___ -- 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
Re: [asterisk-users] Problem with pickup extension *8 from features.conf using IAX
I believe chan_iax2 does not support call pickup. Search the archives.
Shazaum wrote:
already tested with an exten?
ex:
exten = _*8.,1,Pickup(${EXTEN:[EMAIL PROTECTED])
exten = _*8.,n,Hangup()
2008/9/27 coco [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Hello list
I am trying to configure a PBX using Asterisk.
The problem I am havong is the following: I want to use the *8 from
features.conf to pickup any ringing extension from a group, becouse
I want to put the users in call queues and I want anybody from the
company to be able to pick a ringing channel, even if is in a queue.
Whwn using Sip protocol for the users, everithing is going fine, I
can pickup any ringing extension from the group using *8.
But the problem appears when I am using IAX protocol. When issuing
*8 from the IAX phone, asterisk tryes to find the *8 in the dialling
rules returning:
*CLI -- Registered IAX2 '40' (AUTHENTICATED) at 10.0.0.30:4569
http://10.0.0.30:4569
[Sep 27 12:04:33] NOTICE[19796]: chan_iax2.c:8914 socket_process:
Rejected connect attempt from 10.0.0.30 http://10.0.0.30, request
'[EMAIL PROTECTED]' mailto:[EMAIL PROTECTED] does not exist
This I think is wrong, is something like asterisk cannot read from
features.
With the same setting, when using SIP, i get:
*CLI == Using SIP RTP CoS mark 5
[Sep 27 12:06:23] NOTICE[19802]: chan_sip.c:17092
handle_request_invite: Nothing to pick up for
[EMAIL PROTECTED]
mailto:[EMAIL PROTECTED]
and it works ok.
I am wondering if any had this problem before and if you can help me
figure it out(how to make it work--or if is a bug), or find a
sollution using the app pickup.
I tryed using asterisk 1.4.13, asterisk 1.4.21.2 http://1.4.21.2,
asterisk 1.6-rc6 and always the same problem ocurs.
Regards,
Cosmin
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Asterisk user number: 1099
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shazaum.googlepages.com http://shazaum.googlepages.com
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T-1, PRI, Frame Relay, Linux, and network design. Based near
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Re: [asterisk-users] FW: Google Alert - dean collins
Dean Collins wrote: Who is Chris Langford in Huntsville Alabama and is he seeking Digium’s permission in order to report the asterisk mailing lists out onto the internet http://asteriskbizrss.blogspot.com/ http://www.blogger.com/profile/04174728129647374395 What can be done to stop people doing this and making money out of selling ads on these crappy blogsites? Cheers, Dean *From:* Google Alerts [mailto:[EMAIL PROTECTED] *Sent:* Saturday, 27 September 2008 2:38 PM *To:* Dean Collins *Subject:* Google Alert - dean collins Google Blogs Alert for: *dean collins* [asterisk-biz] Philippines http://asteriskbizrss.blogspot.com/2008/09/asterisk-biz-philippines.html By Chris Langford(Chris Langford) I have a friend who is setting up a domestic outbound call center with about 20 agents initially looking for a simple low cost implementation. Email me with reference information and I’ll send you contact details. Regards,. *Dean Collins* *...* Asterisk Biz - http://asteriskbizrss.blogspot.com/ http://asteriskbizrss.blogspot.com/ This as-it-happens Google Alert is brought to you by Google. Remove http://www.google.com/alerts/remove?s=EAWStCkc7lwdmf82QEbYdZMhl=engl= this alert. Create http://www.google.com/alerts?hl=engl= another alert. Manage http://www.google.com/alerts/manage?hl=engl= your alerts. He works for Digium in their sales group. I've known Chris for some time. He's a good guy. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- 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] Troubleshooting one-way voice... how to peek into SIP RTP?
I've got the following situation. I'm running Asterisk 1.4.18 on a firewall/gateway machine, with some SPA-942 (f/w 5.1.15(a)) phones behind it. I'm peering SIP with a Coppercom switch sitting behind an SBC. On outbound calls, I get 2-way voice, no worries. On inbound calls, I get one-way voice (I can hear the caller but they can't hear me). I've looked at tcpdumps of the RTP traffic, and the addresses and port numbers correspond to what's in the SIP INVITE/OK messages (assuming that they don't somehow get munged by NAT after tcpdump looks at them -- there is no NAT device upstream of my Asterisk firewall). I'll look into using Record() or Monitor() to capture the phone call, but if there's any conversion being done by codecs then that won't eliminate the possibility that the code itself is misconfigured or buggy and generating a bad stream on one of the legs... Anyone have an idea about how to best go about troubleshooting this? Thanks, -Philip ___ -- 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
Re: [asterisk-users] FW: Google Alert - dean collins
Pretty easy to do a Google Alert - chris langfordhttp://www.linkedin.com/in/clangford Thanks, Steve Totaro On Sat, Sep 27, 2008 at 5:25 PM, BJ Weschke [EMAIL PROTECTED] wrote: Dean Collins wrote: Who is Chris Langford in Huntsville Alabama and is he seeking Digium's permission in order to report the asterisk mailing lists out onto the internet http://asteriskbizrss.blogspot.com/ http://www.blogger.com/profile/04174728129647374395 What can be done to stop people doing this and making money out of selling ads on these crappy blogsites? Cheers, Dean *From:* Google Alerts [mailto:[EMAIL PROTECTED] *Sent:* Saturday, 27 September 2008 2:38 PM *To:* Dean Collins *Subject:* Google Alert - dean collins Google Blogs Alert for: *dean collins* [asterisk-biz] Philippines http://asteriskbizrss.blogspot.com/2008/09/asterisk-biz-philippines.html By Chris Langford(Chris Langford) I have a friend who is setting up a domestic outbound call center with about 20 agents initially looking for a simple low cost implementation. Email me with reference information and I'll send you contact details. Regards,. *Dean Collins* *...* Asterisk Biz - http://asteriskbizrss.blogspot.com/ http://asteriskbizrss.blogspot.com/ This as-it-happens Google Alert is brought to you by Google. Remove http://www.google.com/alerts/remove?s=EAWStCkc7lwdmf82QEbYdZMhl=engl= this alert. Create http://www.google.com/alerts?hl=engl= another alert. Manage http://www.google.com/alerts/manage?hl=engl= your alerts. He works for Digium in their sales group. I've known Chris for some time. He's a good guy. -- -- Bird's The Word Technologies, Inc. http://www.btwtech.com/ ___ -- 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] New User with Calling Card Question
hi; I'm a new member to this list and have a question for you all. I'm sure it's something simple but alas i must ask. I've wanted to offer calling card features to my customers. For example someone buys a calling card from me for for say 1000 minutes, i give them a phone number/code to call in. However i would like the same number for all callers, just a new card number for different clients. I've looked on google and found a few different things but want to know what you all suggest. I want to get something up maybe in the next 3-5 days. Please email me any ideas. Oh and bandwidth isn't a major issue. It's on a dedicated box with a 100mbps connection to the internet. And finally i will be using completely sip. also, i currently have asterisk installed can i include this calling card in a context in extensions.conf for example: [callingcard] but how would i get my sip file to go to that context when someone new calls in? thanks, sorry for the newbeish questions thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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] Keeps Ringing After Answer
Hi, I searched throug the forum and could not find an answer. Sorry if already posted. I'm using Asterisk 1.4.19 a i686 running Linux Centos. Sometimes when I call and the person picks up, it continues ringing and I can hear the person say hello then it hangs. Has anyone experienced this problem and what is the solution to it. ___ -- 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
Re: [asterisk-users] New User with Calling Card Question
Babcock, Michael Alex wrote: hi; I'm a new member to this list and have a question for you all. I'm sure it's something simple but alas i must ask. I've wanted to offer calling card features to my customers. For example someone buys a calling card from me for for say 1000 minutes, i give them a phone number/code to call in. However i would like the same number for all callers, just a new card number for different clients. I've looked on google and found a few different things but want to know what you all suggest. I want to get something up maybe in the next 3-5 days. Please email me any ideas. Oh and bandwidth isn't a major issue. It's on a dedicated box with a 100mbps connection to the internet. And finally i will be using completely sip. also, i currently have asterisk installed can i include this calling card in a context in extensions.conf for example: [callingcard] but how would i get my sip file to go to that context when someone new calls in? thanks, sorry for the newbeish questions thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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 Hey, You should look into a2billing, its easy to setup, free, and has a bunch of decent features. You can have all the customers call into the same number and just have your extension in the dialplan run DeadAGI(a2billing) and it'll take care of doing the auth/keeping track of how much balance is in the card, etc. a2billing lets you do lcr, multiple ratecards, multiple trunks, etc. Hope it helps, -- Igor Hernandez Escape Communications http://www.escapetel.com ___ -- 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
Re: [asterisk-users] New User with Calling Card Question
You need a billing software or calling card module with an IVR. You can install A2billing in addition to Asterisk. On Sat, Sep 27, 2008 at 6:06 PM, Babcock, Michael Alex [EMAIL PROTECTED]wrote: hi; I'm a new member to this list and have a question for you all. I'm sure it's something simple but alas i must ask. I've wanted to offer calling card features to my customers. For example someone buys a calling card from me for for say 1000 minutes, i give them a phone number/code to call in. However i would like the same number for all callers, just a new card number for different clients. I've looked on google and found a few different things but want to know what you all suggest. I want to get something up maybe in the next 3-5 days. Please email me any ideas. Oh and bandwidth isn't a major issue. It's on a dedicated box with a 100mbps connection to the internet. And finally i will be using completely sip. also, i currently have asterisk installed can i include this calling card in a context in extensions.conf for example: [callingcard] but how would i get my sip file to go to that context when someone new calls in? thanks, sorry for the newbeish questions thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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 ___ -- 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
Re: [asterisk-users] New User with Calling Card Question
can a2 billing work on the same system that directadmin is installed? On Sep 27, 2008, at 2:21 PM, broadband Voice wrote: You need a billing software or calling card module with an IVR. You can install A2billing in addition to Asterisk. On Sat, Sep 27, 2008 at 6:06 PM, Babcock, Michael Alex [EMAIL PROTECTED] wrote: hi; I'm a new member to this list and have a question for you all. I'm sure it's something simple but alas i must ask. I've wanted to offer calling card features to my customers. For example someone buys a calling card from me for for say 1000 minutes, i give them a phone number/code to call in. However i would like the same number for all callers, just a new card number for different clients. I've looked on google and found a few different things but want to know what you all suggest. I want to get something up maybe in the next 3-5 days. Please email me any ideas. Oh and bandwidth isn't a major issue. It's on a dedicated box with a 100mbps connection to the internet. And finally i will be using completely sip. also, i currently have asterisk installed can i include this calling card in a context in extensions.conf for example: [callingcard] but how would i get my sip file to go to that context when someone new calls in? thanks, sorry for the newbeish questions thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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 ___ -- 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 thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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] credit card processing
Hi Guys We have a service that can be use by our customer via a website and also via telephone. On the website, we already accept credit card by sending users to paypal website where we have an account. Now, we want to do the same with an IVR where people can call a number, enter their credit card number and expiration date. But I don't see any service or credit card procession company that offers this. What I want basicly is a service where I can send the credit card number I collected and expiration that and their charge the number and give me a status back. Do you know any company that do this ?? Thanks ___ -- 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
Re: [asterisk-users] credit card processing
I might want something like this to, hmm. On Sep 27, 2008, at 2:52 PM, Ruddy Gbaguidi wrote: Hi Guys We have a service that can be use by our customer via a website and also via telephone. On the website, we already accept credit card by sending users to paypal website where we have an account. Now, we want to do the same with an IVR where people can call a number, enter their credit card number and expiration date. But I don't see any service or credit card procession company that offers this. What I want basicly is a service where I can send the credit card number I collected and expiration that and their charge the number and give me a status back. Do you know any company that do this ?? Thanks ___ -- 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 thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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
Re: [asterisk-users] credit card processing
Most credit card processing gateways require you to have the user's name and address for AVS verification when you perform customer not present transactions. Easy enough to do over a website, but a bit more tricky on the phone. If these are for repeat orders, how about getting the user to register via the website first, entering a payment card to be used for future orders, then give then a customer number and PIN that can be used by telephone for future top-up orders? Something like that would be fairly easy to query against a database. I'd have thought. Regards, Chris ___ -- 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
Re: [asterisk-users] credit card processing
that would work for what i might use it for but not required right now... On Sep 27, 2008, at 3:16 PM, Chris Bagnall wrote: Most credit card processing gateways require you to have the user's name and address for AVS verification when you perform customer not present transactions. Easy enough to do over a website, but a bit more tricky on the phone. If these are for repeat orders, how about getting the user to register via the website first, entering a payment card to be used for future orders, then give then a customer number and PIN that can be used by telephone for future top-up orders? Something like that would be fairly easy to query against a database. I'd have thought. Regards, Chris ___ -- 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 thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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
Re: [asterisk-users] credit card processing
Yes, we can do that. But : 1. we are not too confortabe about keeping users credit card informations in our databases 2. we are now targeting the 50, 60+ people and their are not confortable about a website. So, we want to be able to register people by phone, and they can make payments by phone. We provide long distance service, so the website is only for payments for now. It will be more easier if people can pay by phone as well. Chris Bagnall wrote: Most credit card processing gateways require you to have the user's name and address for AVS verification when you perform customer not present transactions. Easy enough to do over a website, but a bit more tricky on the phone. If these are for repeat orders, how about getting the user to register via the website first, entering a payment card to be used for future orders, then give then a customer number and PIN that can be used by telephone for future top-up orders? Something like that would be fairly easy to query against a database. I'd have thought. Regards, Chris ___ -- 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 Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- 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
Re: [asterisk-users] test call generator
I've used both the Hammer Call Analyzer software and als to the Hammer XMS system which is a server that they install in your rack to do the packet captures and provide you with all sorts of statistics. I suspect the Empirix Hammer products would be able to take care of any load, monitoring or analysis scenarios you have including signalling and media. The price is going to be the issue. When we looked at the solution the Call Analyzer software was 5 figures and the XMS solution was 6. Regards, Greyman. ___ -- 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] G.722 between Eyebeam and a Polycom IP650
Hi All, So I've been exploring the use of G.722 encoded wideband audio recently. I have three different SIP devices that allow this: Eyebeam, IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine together. Calls pass between them in what the Polycom notes as HD mode and the audio quality is certainly very good. However, things are not so easy with Eyebeam and the IP650. When a call is placed between those two the audio stream from Eyebeam to the IP650 is never heard. The audio from the IP650 to Eyebeam is heard, and very good quality. David Frankel of ZipDX tells me that there is an error in RFC3551 such that G.722 RTP clock/timestamps are actually wrong. To quote the RFC directly. Even though the actual sampling rate for G.722 audio is 16,000 Hz, the RTP clock rate for the G722 payload format is 8,000 Hz because that value was erroneously assigned in RFC 1890 and must remain unchanged for backward compatibility. The octet rate or sample-pair rate is 8,000 Hz. It seems that some manufacturers adhere strictly to the RFC while others correct for the error. As such there are problems with G.722 interoperability. Counterpath defends their implementation as being according to the RFC. This begs the question of what does Asterisk do with G.722? I've yet to try v1.6 so I open the question to the group. Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- 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
Re: [asterisk-users] credit card processing
or what about if you took credit card, and billing zip code would there be any processors that would let you do that, maybe the 3-4 digit security code on the back of the card? mike with just some ideas. On Sep 27, 2008, at 4:14 PM, Ruddy Gbaguidi wrote: Yes, we can do that. But : 1. we are not too confortabe about keeping users credit card informations in our databases 2. we are now targeting the 50, 60+ people and their are not confortable about a website. So, we want to be able to register people by phone, and they can make payments by phone. We provide long distance service, so the website is only for payments for now. It will be more easier if people can pay by phone as well. Chris Bagnall wrote: Most credit card processing gateways require you to have the user's name and address for AVS verification when you perform customer not present transactions. Easy enough to do over a website, but a bit more tricky on the phone. If these are for repeat orders, how about getting the user to register via the website first, entering a payment card to be used for future orders, then give then a customer number and PIN that can be used by telephone for future top-up orders? Something like that would be fairly easy to query against a database. I'd have thought. Regards, Chris ___ -- 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 Internal Virus Database is out of date. Checked by AVG. Version: 8.0.100 / Virus Database: 269.23.16/1448 - Release Date: 5/16/2008 7:42 PM ___ -- 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 thanks for reading Systems administrator and owner of http://gwhosting.net msn: [EMAIL PROTECTED] twitter: http://twitter.com/creepyblindy ___ -- 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] Eye P Media Soft Phone?
Does anyone on the list have any experience with this soft phone? Thanks, Michael -- Michael Graves mgravesatmstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 ___ -- 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] Vividial issue
does anyone have a sample dialplan for vici dial that does not include any pri stuff. I am running exclusively SIP for everything and trying to edit the sample dialplan and removing anything to do with a pri card is becoming a nightmare! Thank you! ___ -- 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
Re: [asterisk-users] G.722 between Eyebeam and a Polycom IP650
Michael Graves wrote: Hi All, So I've been exploring the use of G.722 encoded wideband audio recently. I have three different SIP devices that allow this: Eyebeam, IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine together. Calls pass between them in what the Polycom notes as HD mode and the audio quality is certainly very good. However, things are not so easy with Eyebeam and the IP650. When a call is placed between those two the audio stream from Eyebeam to the IP650 is never heard. The audio from the IP650 to Eyebeam is heard, and very good quality. David Frankel of ZipDX tells me that there is an error in RFC3551 such that G.722 RTP clock/timestamps are actually wrong. To quote the RFC directly. Even though the actual sampling rate for G.722 audio is 16,000 Hz, the RTP clock rate for the G722 payload format is 8,000 Hz because that value was erroneously assigned in RFC 1890 and must remain unchanged for backward compatibility. The octet rate or sample-pair rate is 8,000 Hz. It seems that some manufacturers adhere strictly to the RFC while others correct for the error. As such there are problems with G.722 interoperability. Counterpath defends their implementation as being according to the RFC. This begs the question of what does Asterisk do with G.722? I've yet to try v1.6 so I open the question to the group. Michael There is no error in RFC3551. There is a clear statement that an earlier RFC did things wrong, due to a typo, and classifying G.722 as an 8000 sample/second codec is, for better or worse, the standard. Its messy and inconsistent, but its the standard. The only manufacturers I know of who do the wrong thing (i.e. using 16000 in the SDP) are Grandstream and Aastra. Both are aware that their products are incompatible with the rest of the universe, but seem uninterested in fixing them. If I were building a terminal, I'd make mine announce 8000, but accept 8000 or 16000 to try to maximise compatibility. It seems people don't do that. Unless you have some special version of eyebeam, I don't think it supports G.722. It supports G.722.2, but that is completely different. It also supports 16 bit PCM and DVI4 at 16000 samples/second. Regards, Steve ___ -- 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
