Re: [asterisk-users] Fax with Asterisk + Hylafax
From: Tzafrir Cohen [EMAIL PROTECTED] Date: Mon, 16 Apr 2007 08:45:38 +0300 On Sun, Apr 15, 2007 at 10:10:34PM -0700, Yuan LIU wrote: (But if Zaptel and Hylafax can share an X100P driver ...) Where can you find a modem driver for a X100P? Kinda my question, too. Motorola used to have an SM56 Linux driver, but removed from their site. Now, there are some references to this, such as http://www.motorola.com/softmodem/public_download/Linux/ReadMe_Legacy_SM56.txt and http://www.angelfire.com/linux/sm56/, but if the original driver is nowhere to be downloaded, there might be a chance you can hack the URL based on the Motorola document. No knowledge about X100P/Intel and other. Yuan Liu I recently asked about it in the linmodemds.org mailing list, and aparantly none is available. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Fax with Asterisk + Hylafax
Steve Totaro wrote: Stephen Bosch wrote: Steve Totaro wrote: You could try to get it working but it may never be 100%. If your needs are 100% then I suggest using a standard fax and get an analog line and do it the old fashioned way. If you need Hylafax type features then buy a modem that is compatible with Hylafax and run it on a different box. It's not entirely clear to me why people continue to cling to the idea that Asterisk should handle faxing also. What's the benefit? Hylafax is great, and you can even use it on the same machine. -Stephen- I could have sworn that is what I just said. Yes, and I was commenting :) -Stephen- ___ --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] Fax with Asterisk + Hylafax
My problem is that I already did what you proposed and I did not have much sucess. An Hylafax server with an external/internal modem using a driver from linmodems worked in around 80% of the cases. To improve this rate you need a modem like the ones offered by Mainpine which were way out of my budget. Regards, Jose L. On 16/04/07, Tzafrir Cohen [EMAIL PROTECTED] wrote: On Sun, Apr 15, 2007 at 10:10:34PM -0700, Yuan LIU wrote: (But if Zaptel and Hylafax can share an X100P driver ...) Where can you find a modem driver for a X100P? I recently asked about it in the linmodemds.org mailing list, and aparantly none is available. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Is STP wire decent for analog phones?
On Sun, 15 Apr 2007, Steve Prior wrote: I've got a run of Shielded Twisted Pair (4 conductors) which used to be a Token Ring Network drop and I'm not using it anymore. Would it be decent to replace the ends with normal analog phone connectors and use it for a phone extension, or is STP unsuitable for that? I've always reckoned you can get away with just about anything for analogue phones - especially internal wiring. After-all, a POTS line from the exchange has just come over who knows how many miles of aged copper wires, so a few meters more in the building of reasonable quality cable isn't going to hurt it! (and if you're re-originating it from an in-house PBX then you're starting with a near perfect signal again) What I would pay attention to is the connectors though - I've seen these cause more problems - especially for ADSL lines... Gordon ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] New T1 Asterisk installation
Hi List, I need to change my provider, at this time Asterisk box is on VOIP trunk. I have two options, T1 or 15 analog lines. I have some experience with analog and I have had two main issues with it. first is echo (I have not tried HPEC yet) and second unpredictable volume. The question is, if I use TE100 with PRI , will I have same issues? I would appreciate any comments and sample zaptel.conf and zapata.conf___ --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] New T1 Asterisk installation
In general, you're going to have better luck with a PRI. A lot of echo occurs at the point where analog lines are broken out of a digital transport. Also, from an economic standpoint, you *really*, *really* don't want to order 15 analog lines. Generally the rule of thumb in most areas of the US is that if you're ordering six or more analog lines, you might as well just buy a PRI, the way the price points are structured. Plus, with 15 analog lines, that's a lot of termination hardware. And if you're thinking of putting it in the same chassis, expect possible bus problems / IRQ issues / etc. of the sort that are regularly discussed here. Sorry if I misunderstood your question... -- Alex On Mon, 16 Apr 2007, Al said something to this effect: Hi List, I need to change my provider, at this time Asterisk box is on VOIP trunk. I have two options, T1 or 15 analog lines. I have some experience with analog and I have had two main issues with it. first is echo (I have not tried HPEC yet) and second unpredictable volume. The question is, if I use TE100 with PRI , will I have same issues? I would appreciate any comments and sample zaptel.conf and zapata.conf -- Alex Balashov [EMAIL PROTECTED] ___ --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] Debian asterisk-bristuff
I apologise now if I have managed to completely misunderstand this whole subject! I've built a small PC and loaded Etch 4.0 from the netinst cd. I did 'apt-get install asterisk-bristuff' which seemed to work but, it doesn't seem to have installed any files/modules for zaptel? ztcfg zaptel zaphfc I am using a billion hfc card Any pointers? -- Simon Faulkner 01538 303 900 Staffordshire Moorlands ___ --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] MySQL query from extensions?
I also dropped the quotes on the dnis=${IVR-Exten}.
That's only allowed if the dnis column doesn't contain a string.
--
Andreas SikkemaBBeyond
Software EngineerPlaneetbaan 4
+31 (0)23 70743422132 HZ Hoofddorp
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Re: [asterisk-users] agents and music on hold with autoanswer..
BUT if the agent have to go elsewhere for some minutes (coffe break, go to piss, and so on..), usually he press the 'hold' button on the phone; Does the phone have a DND (Do not disturb) button? yes, the phone have this options; I have to check if that works Are all the agents trained to press hold when they need to go the bathroom? yes... If the answer to the last question is yes, you have more than a technology problem on your hands. Perhaps this is why your colleague left in the first place :) may be you're right :) We are using asterisk 1.2.1 with Thomson ST2030. The Thomson is the telephone set? yes. and I use the 'diva' modules for the phone card on asterisk (...) -- Executing Queue(CAPI/ISDN4/-ce, coda_azienda|t| 3600) in new stack -- Started music on hold, class 'music', on channel 'CAPI/ISDN4/**-ce' -- agent_call, call to agent '1005' call on 'SIP/barbaran-621c' -- Playing 'beep' (language 'it') -- Called Agent/1005 -- Agent/1005 answered CAPI/ISDN4/**-ce [WARNING: in the truth the Agent is in hold mode now; there is the autoanswer on] (!) Why? (FYI: Auto answer is normally enabled in the telephone configuration and not in Asterisk.) we don't receive so many calls/minute; usually the agent keep the own phone with the speaker on, to have the hands free; when there is a new call on the queue, he can hear the voice (a new call is incoming), then he press the 'speaker off' button and the handle the call normally... then the call is finished, he doesn't hangup the phone, but he puts again the phone with the speaker on [AND NOW THE CALLER DON'T HEAR ANYTHINGuntil the agent will press the hold button again] Well, that's to be expected. The phone has answered the call! true :( A few bits of advice to start: 1. Agents shouldn't be using hold for bathroom breaks. Most phones have a button specifically for this purpose called Do not disturb. Asterisk then treats the station as busy. I have to try that... 2. Queue phones shouldn't answer automatically. That's just inviting disaster. What if somebody forgets to log out when they leave? Somebody is going to get silence if they're unlucky enough to be connected to that agent. you're right.. I have to investigate here too! Fix both those things and you won't have to worry about Music On Hold not playing for the caller :) :) ok, thank you so much! bye bye marco ___ --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] Loudspeaker
Just run down to your local Radio Shack...and KISS. http://www.radioshack.com/product/index.jsp?productId=2062696 Mark C. Klaverstyn, David C wrote: This is what I want. Do you have any URLs to such a device as I cannot find any. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of cb Sent: Monday, 16 April 2007 12:24 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Loudspeaker On Apr 15, 2007, at 9:53 PM, Klaverstyn, David C wrote: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? Do you already have the loud speaker? If not, I know there are various vendors of extension phone bells that do nothing more than plug into an analog line and ring the nice loud bell when a ring signal is received. You could easily combine one of those with a cheap ATA with FXS port. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Debian asterisk-bristuff
On Mon, Apr 16, 2007 at 08:49:18AM +0100, Simon Faulkner wrote: I apologise now if I have managed to completely misunderstand this whole subject! I've built a small PC and loaded Etch 4.0 from the netinst cd. I did 'apt-get install asterisk-bristuff' which seemed to work but, it doesn't seem to have installed any files/modules for zaptel? ztcfg zaptel zaphfc I am using a billion hfc card apt-get install zaptel-source m-a a-i zaptel Precompiled zaptel drivers should hopefully be added soon to Unstable / Testing . -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] injecting audio announcements into sip channel
Hi all, I have a client that is setting up a premium phone service and is required by the Telephony regulator to stream call cost announcements into the call at certain periods during the call. For instance when the call cost is €3 per minute there needs to be an announcement after 10 minutes that call has now cost you €30 to the caller. €60 after 20 mins etc. etc. Recording the bespoke announcements is easy enough. 1) But how do I inject them into the SIP channel. 2) How do I time the injection so that the correct message is played at the correct time. I imagine there is a more elegant way to do this than setting up a conference call and have the announcements played as part of one long recording on one of the conference channels. Advice from anyone who has tackled this problem would be greatly appreciated. cheers, Mark ___ --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] injecting audio announcements into sip channel
On Mon, 16 Apr 2007 10:48:40 +0100, Mark Reardon wrote: 1) But how do I inject them into the SIP channel. 2) How do I time the injection so that the correct message is played at the correct time. take a look at the L() option to Dial(). -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sip tcp support
Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose is ICMP Destination unreachable (Port unreachable). any hints? Thx in advance Xtenasterisk HiPath | INVITE| | |--| | | TRYING| | |--| | | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| |603 DECINE | | |--| | | ACK | | |--| | - Registered SIP '971' at 10.4.5.1 port 5060 expires 120 proxy*CLI sip show peer 971 * Name : 971 Secret : Not set MD5Secret: Not set Context : from_hipath Subscr.Cont. : Not set Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : Expire : 113 Insecure : no Nat : RFC3581 ACL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 10.4.5.1 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Sock fd : 24 Transport: TCP Def. Username: 971 SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order : (alaw,ulaw) Status : Unmonitored Useragent: HiPath 4000 V3.0 M5T SIP-UA SAFE/v3.6.6.10 Reg. Contact : sip:[EMAIL PROTECTED]:5060;transport=tcp __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip tcp support
richard Coco wrote: Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose is ICMP Destination unreachable (Port unreachable). any hints? Thx in advance Xtenasterisk HiPath | INVITE| | |--|| | TRYING| | |--|| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| |603 DECINE | | |--|| | ACK | | |--|| - Registered SIP '971' at 10.4.5.1 port 5060 expires 120 proxy*CLI sip show peer 971 * Name : 971 Secret : Not set MD5Secret: Not set Context : from_hipath Subscr.Cont. : Not set Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : Expire : 113 Insecure : no Nat : RFC3581 ACL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 10.4.5.1 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Sock fd : 24 Transport: TCP Def. Username: 971 SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order : (alaw,ulaw) Status : Unmonitored Useragent: HiPath 4000 V3.0 M5T SIP-UA SAFE/v3.6.6.10 Reg. Contact : sip:[EMAIL PROTECTED]:5060;transport=tcp __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Just out of sheer curiousity, I'm wondering why you decided to use TCP as opposed to UDP. Please don't tell me its for security reasons... Just a question. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip tcp support
On Mon, Apr 16, 2007 at 03:03:30AM -0700, richard Coco wrote: Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose is ICMP Destination unreachable (Port unreachable). Anybody listens on the UDP port? netstat -lnup | grep 5060 any hints? Thx in advance Xten asterisk HiPath | INVITE| | |--| | | TRYING| | |--| | | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| |603 DECINE | | |--| | | ACK | | |--| | - Registered SIP '971' at 10.4.5.1 port 5060 expires 120 Should that message be changed to reflect the fact that the port is TCP? (and is it for a TCP port indeed?) proxy*CLI sip show peer 971 * Name : 971 Secret : Not set MD5Secret: Not set Context : from_hipath Subscr.Cont. : Not set Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : Expire : 113 Insecure : no Nat : RFC3581 ACL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 10.4.5.1 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Sock fd : 24 Transport: TCP Def. Username: 971 SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order : (alaw,ulaw) Status : Unmonitored Useragent: HiPath 4000 V3.0 M5T SIP-UA SAFE/v3.6.6.10 Reg. Contact : sip:[EMAIL PROTECTED]:5060;transport=tcp -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip tcp support
--- J. Oquendo [EMAIL PROTECTED] wrote: richard Coco wrote: Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose is ICMP Destination unreachable (Port unreachable). any hints? Thx in advance Xtenasterisk HiPath | INVITE| | |--| | | TRYING| | |--| | | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| |603 DECINE | | |--| | | ACK | | |--| | - Registered SIP '971' at 10.4.5.1 port 5060 expires 120 proxy*CLI sip show peer 971 * Name : 971 Secret : Not set MD5Secret: Not set Context : from_hipath Subscr.Cont. : Not set Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : Expire : 113 Insecure : no Nat : RFC3581 ACL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 10.4.5.1 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Sock fd : 24 Transport: TCP Def. Username: 971 SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order : (alaw,ulaw) Status : Unmonitored Useragent: HiPath 4000 V3.0 M5T SIP-UA SAFE/v3.6.6.10 Reg. Contact : sip:[EMAIL PROTECTED]:5060;transport=tcp __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: i have read somewhere that the HG3540 only works with sip tcp for SIPQ. http://lists.digium.com/mailman/listinfo/asterisk-users Just out of sheer curiousity, I'm wondering why you decided to use TCP as opposed to UDP. Please don't tell me its for security reasons... Just a question. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --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 __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] injecting audio announcements into sip channel
--- Dinesh Nair [EMAIL PROTECTED] wrote: take a look at the L() option to Dial(). The original poster said he need to play different messages at different call durations. In order to do that you would need to dynamically alter LIMIT_WARNING_FILE as the call progressed. Regards Jon Jon Farmer Telford, Shropshire, UK ___ Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today http://uk.rd.yahoo.com/evt=44106/*http://uk.docs.yahoo.com/mail/winter07.html ___ --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] sip tcp support
strange i have: udp0 0 0.0.0.0:5060 0.0.0.0:* 9722/asterisk 972 is the tie access code from Hiapth to Asterisk. --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Apr 16, 2007 at 03:03:30AM -0700, richard Coco wrote: Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose is ICMP Destination unreachable (Port unreachable). Anybody listens on the UDP port? netstat -lnup | grep 5060 any hints? Thx in advance Xtenasterisk HiPath | INVITE| | |--| | | TRYING| | |--| | | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| |603 DECINE | | |--| | | ACK | | |--| | - Registered SIP '971' at 10.4.5.1 port 5060 expires 120 Should that message be changed to reflect the fact that the port is TCP? (and is it for a TCP port indeed?) proxy*CLI sip show peer 971 * Name : 971 Secret : Not set MD5Secret: Not set Context : from_hipath Subscr.Cont. : Not set Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : Expire : 113 Insecure : no Nat : RFC3581 ACL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 10.4.5.1 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Sock fd : 24 Transport: TCP Def. Username: 971 SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order : (alaw,ulaw) Status : Unmonitored Useragent: HiPath 4000 V3.0 M5T SIP-UA SAFE/v3.6.6.10 Reg. Contact : sip:[EMAIL PROTECTED]:5060;transport=tcp -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Queue trouble
Hi everyone, I'm in trouble with queue. There are a little local radio station with one studio and we have to switch queued callers to the live program. Everything works fine (counting callers, periodic announcements), but while the announcement is played for 'firs in line' caller, studio gets a free line out not the caller. member = SIP/suich ;only 1 member strategy = ringall Asterisk 1.2.17 built by root Any idea what can I do with that? -- Suich ___ --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] Instability on Asterisk
Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:19] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) Sniffing netowork i saw register packets arriving from ip phones, but asterisk didn't send response to it, and for external registry i saw register sended to sip provider, 401 response from sip provider and asterisk didn't start sip digest challenger, it was send a register message again without authentication header. Network connectivity for asterisk was ok during this problem moments. My asterisk is 1.4.2 with FC6 What can be wrong ? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --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] injecting audio announcements into sip channel
Thanks. On 4/16/07, Dinesh Nair [EMAIL PROTECTED] wrote: On Mon, 16 Apr 2007 10:48:40 +0100, Mark Reardon wrote: 1) But how do I inject them into the SIP channel. 2) How do I time the injection so that the correct message is played at the correct time. take a look at the L() option to Dial(). -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] injecting audio announcements into sip channel
yeah, it would be good to have 2 different messages. But I guess I could adapt the message to be generic enough to cover both scenarios - your call has now cost over €X. This might scrape through with legal requirements. cheers for the feedback - I appreciate it. On 4/16/07, Jon Farmer [EMAIL PROTECTED] wrote: --- Dinesh Nair [EMAIL PROTECTED] wrote: take a look at the L() option to Dial(). The original poster said he need to play different messages at different call durations. In order to do that you would need to dynamically alter LIMIT_WARNING_FILE as the call progressed. Regards Jon Jon Farmer Telford, Shropshire, UK ___ Yahoo! Mail is the world's favourite email. Don't settle for less, sign up for your free account today http://uk.rd.yahoo.com/evt=44106/*http://uk.docs.yahoo.com/mail/winter07.html ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Instability on Asterisk
Frederico Madeira wrote: Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:19] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) Sniffing netowork i saw register packets arriving from ip phones, but asterisk didn't send response to it, and for external registry i saw register sended to sip provider, 401 response from sip provider and asterisk didn't start sip digest challenger, it was send a register message again without authentication header. Network connectivity for asterisk was ok during this problem moments. My asterisk is 1.4.2 with FC6 What can be wrong ? Thanks. Why do you believe this to be an issue with Asterisk. What you describe is this in barebones YourNetworkPhones --- connect --- SIP Provider SIP Provider --- starts handshake --- Yournetwork YourNetwork --- gets ball rolling --- SIP Provider SIP Provider ... ignores you Sounds like you should be ripping into your SIP provider they're sending you unauthorized messages which sounds like either they changed something, or you did. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Difference between SCCP and Cisco Call Manager traffic?
I'm wondering about the difference between Cisco Call Manager and SCCP(2) network traffic. I'm working on getting a Cisco 7960 phone to speak through a NAT to an asterisk box, without having to do a bunch of port forwarding on the NAT device. Without the nat, everything works fine. If the phone is behind a cisco pix that is doing the natting, it works fine (fixup protocol). If the phone is behind a more generic nat device, such as a linux box running ipfilter. Then it can dial out, but there is no audio. The interesting part is that this same phone, behind the same NAT works just fine if it is talking to a Cisco Call Manager box instead of an asterisk server. So, I'm wondering what the difference in the protocols is (I no longer have access to the call manager box, so I can't look @ the traffic). In a perfect world, I'd like to have the phone pretty much just work wherever it's plugged in as long as it can see the asterisk server. Any ideas ? Thanks Shawn ___ --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] Debian asterisk-bristuff
I am using a billion hfc card apt-get install zaptel-source m-a a-i zaptel Precompiled zaptel drivers should hopefully be added soon to Unstable / Testing . Thank you :-) ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sip tcp support
sorry, it works with upd... I am now able to make and to receive calls. thx... --- richard Coco [EMAIL PROTECTED] wrote: strange i have: udp0 0 0.0.0.0:5060 0.0.0.0:* 9722/asterisk 972 is the tie access code from Hiapth to Asterisk. --- Tzafrir Cohen [EMAIL PROTECTED] wrote: On Mon, Apr 16, 2007 at 03:03:30AM -0700, richard Coco wrote: Hi all, i have asterisk 1.2.17 with sip tcp support and i am trying to connect asterisk with HiPath 4000 V.3.0 using SIP. I can see the registration from the HG3540. But when i try to place a call from Asterisk to HiPath, the call fails with SIP/2.0 603 Declined. The strange thing is that the first INVITE uses tcp and the response is a 100 TRYING, the next 7 INVITE are using udp and the respose is ICMP Destination unreachable (Port unreachable). Anybody listens on the UDP port? netstat -lnup | grep 5060 any hints? Thx in advance Xten asterisk HiPath | INVITE| | |--| | | TRYING| | |--| | | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| | | INVITE| | |--| |603 DECINE | | |--| | | ACK | | |--| | - Registered SIP '971' at 10.4.5.1 port 5060 expires 120 Should that message be changed to reflect the fact that the port is TCP? (and is it for a TCP port indeed?) proxy*CLI sip show peer 971 * Name : 971 Secret : Not set MD5Secret: Not set Context : from_hipath Subscr.Cont. : Not set Language : AMA flags: Unknown CallingPres : Presentation Allowed, Not Screened Callgroup: Pickupgroup : Mailbox : VM Extension : asterisk LastMsgsSent : 32767/65535 Call limit : 0 Dynamic : Yes Callerid : Expire : 113 Insecure : no Nat : RFC3581 ACL : No CanReinvite : Yes PromiscRedir : No User=Phone : No Trust RPID : No Send RPID: No DTMFmode : rfc2833 LastMsg : 0 ToHost : Addr-IP : 10.4.5.1 Port 5060 Defaddr-IP : 0.0.0.0 Port 5060 Sock fd : 24 Transport: TCP Def. Username: 971 SIP Options : (none) Codecs : 0xc (ulaw|alaw) Codec Order : (alaw,ulaw) Status : Unmonitored Useragent: HiPath 4000 V3.0 M5T SIP-UA SAFE/v3.6.6.10 Reg. Contact : sip:[EMAIL PROTECTED]:5060;transport=tcp -- 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 Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users __ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Dial n voicemaile
Hi, While Dial rings can a caller press 0 (or other number) to leave a voicemail? I found that with a # can transfer to different context. I want to use that two features together. -- Suich ___ --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] Difference between SCCP and Cisco Call Manager traffic?
Call setup/teardown is handled with the SIP protocol while the actual call audio is handled with RTP I think. Check the config of your NAT devices relative to RTP. scd On 4/16/07, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote: I'm wondering about the difference between Cisco Call Manager and SCCP(2) network traffic. I'm working on getting a Cisco 7960 phone to speak through a NAT to an asterisk box, without having to do a bunch of port forwarding on the NAT device. Without the nat, everything works fine. If the phone is behind a cisco pix that is doing the natting, it works fine (fixup protocol). If the phone is behind a more generic nat device, such as a linux box running ipfilter. Then it can dial out, but there is no audio. The interesting part is that this same phone, behind the same NAT works just fine if it is talking to a Cisco Call Manager box instead of an asterisk server. So, I'm wondering what the difference in the protocols is (I no longer have access to the call manager box, so I can't look @ the traffic). In a perfect world, I'd like to have the phone pretty much just work wherever it's plugged in as long as it can see the asterisk server. Any ideas ? Thanks Shawn ___ --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 -- Steve Dickey Who is John Galt? ___ --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] Which SIP phones to buy?
Luca Corti wrote: On Fri, 2007-04-13 at 17:46 +0200, map wrote: Linksys SPAs work well with Asterisk I know, I use them and besides some initial nasty bugs and occasional quirks they are quite nice. I also think they are not so ugly. Luca, what sort of nasty bugs and quirks have you seen with the Linksys SPA? We've recently started using a few SPA-921, and will probably be buying some more. /Per Jessen, Zürich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] zaptel/ssh interaction (SOLVED!)
On Sun, 2007-04-15 at 14:53 -0600, Greg Woods wrote: when I recompiled zaptel with 1.4.1 and installed that, the problem is gone. I don't know if this was due to changes I made in the 1.4.0 zconfig.h file, or that there were fixes in 1.4.1. I checked, and the zconfig.h file that is in my 1.4.0 directory is identical with the one from the 1.4.1 directory, so the problem is unlikely to have been caused by changes I made. --Greg ___ --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] agents and music on hold with autoanswer..
If you want to be able to run accurate reporting, you should tell the agents that they must log out whenever they are unavailable to answer calls. If accurate reporting is something you may be interested in doing in the future, then make this the rule now. Autoanswering queues is great for productivity. There are ways to implement an auto-logout due to inactivity if you want to continue using this strategy. I always thought if there were a reverse answering machine detection type of application, where if there is silence on the agent's leg of the call, the system would log them out, this would be ideal. I just never got around to making that happen. What I have used is that if there is no activity on the workstation, similar to a screensaver kicking in, the user is logged out of queue. In the environments that I have worked in that dealt with queues and agents, they are a sneaky bunch. Strict rules must be defined and enforced. I have seen operations where no policing of agents resulted in agents calling their friends all day and collecting a paycheck at the end of the pay period. I have also seen them use every possible technique to be logged in but not have to receive calls. If it is possible, they will figure it out. Thanks, Steve Totaro http://www.asteriskhelpdesk.com KB3OPB -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of MAS! Sent: Monday, April 16, 2007 5:00 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] agents and music on hold with autoanswer.. BUT if the agent have to go elsewhere for some minutes (coffe break, go to piss, and so on..), usually he press the 'hold' button on the phone; Does the phone have a DND (Do not disturb) button? yes, the phone have this options; I have to check if that works Are all the agents trained to press hold when they need to go the bathroom? yes... If the answer to the last question is yes, you have more than a technology problem on your hands. Perhaps this is why your colleague left in the first place :) may be you're right :) We are using asterisk 1.2.1 with Thomson ST2030. The Thomson is the telephone set? yes. and I use the 'diva' modules for the phone card on asterisk (...) -- Executing Queue(CAPI/ISDN4/-ce, coda_azienda|t| 3600) in new stack -- Started music on hold, class 'music', on channel 'CAPI/ISDN4/**-ce' -- agent_call, call to agent '1005' call on 'SIP/barbaran-621c' -- Playing 'beep' (language 'it') -- Called Agent/1005 -- Agent/1005 answered CAPI/ISDN4/**-ce [WARNING: in the truth the Agent is in hold mode now; there is the autoanswer on] (!) Why? (FYI: Auto answer is normally enabled in the telephone configuration and not in Asterisk.) we don't receive so many calls/minute; usually the agent keep the own phone with the speaker on, to have the hands free; when there is a new call on the queue, he can hear the voice (a new call is incoming), then he press the 'speaker off' button and the handle the call normally... then the call is finished, he doesn't hangup the phone, but he puts again the phone with the speaker on [AND NOW THE CALLER DON'T HEAR ANYTHINGuntil the agent will press the hold button again] Well, that's to be expected. The phone has answered the call! true :( A few bits of advice to start: 1. Agents shouldn't be using hold for bathroom breaks. Most phones have a button specifically for this purpose called Do not disturb. Asterisk then treats the station as busy. I have to try that... 2. Queue phones shouldn't answer automatically. That's just inviting disaster. What if somebody forgets to log out when they leave? Somebody is going to get silence if they're unlucky enough to be connected to that agent. you're right.. I have to investigate here too! Fix both those things and you won't have to worry about Music On Hold not playing for the caller :) :) ok, thank you so much! bye bye marco ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Hardware requirements question
On Saturday 14 April 2007 00:52, [EMAIL PROTECTED] wrote: Can you tell me if this sounds sane? We are planning on using a Dell 933Mhz dual CPU server, with 1GB of ram for our Trixbox setup. We will have 7-10 internal phones, and maybe 3-4 max outbound connections at a time. We will have some type of menu system for inbound callers. At this point I'm planning on connecting to a SIP provider over the internet for service. Do you think the hardware is adequate? If there's a chance its not enough horsepower I want to find a different server. I'm not an expert, but I'd say that this is pretty much spot-on for what you're trying to do. We've deployed systems before with twice the number of extensions and half the horsepower with no problems. -- Charles Ulrich Ideal Solution, LLC -- http://www.idealso.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Re: New T1 Asterisk installation
Quoting [EMAIL PROTECTED]: I have two options, T1 or 15 analog lines. The question is, if I use TE100 with PRI , will I have same issues? I would appreciate any comments and sample zaptel.conf and zapata.conf 15 lines should be well beyond the cost justification point for a T1 and you will get significantly better quality (disconnects) and functionality out of digital trunk. Plus you clean up the telco closet. Remember, you don't need to activate all 23 lines so if you just need 15 then you can activate only that number. You also can have potentially hundreds of numbers that terminate on this group of lines. This makes some of your coding a little different because you no longer make an association between which port(s) ring and what number the caller dialed to get here. This is called DNIS (Dialed Number Identification System) (people don't flame me for the ANI/DNIS thing OK? Not relevant for this discussion). When ordering the PRI the telco will ask you what type of signaling you want and how many DNIS digits. Personally, as we have intermixed area codes, I always ask for 10 digits DNIS. This means when asterisk answers the phone the $EXTEN will equal the full phone number the caller dialed to get here. loadzone= us defaultzone = us span= 1,2,0,esf,b8zs bchan = 01-23 dchan = 24 span= 2,3,0,esf,b8zs bchan = 25-30 dchan = 48 This is a zaptel.conf for 2 PRI's. 23 chan on 1 and 6 on the second. It stipulates ESF (Extended SuperFrame) with b8zs coding. Both PRI's have their D channel on the last (24th) channel. As for emulation I try to ask for NI2 (which is a config that goes in zapata.conf for switchtype). Hope this gets you started. -- dbc. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unicall/R2 for Asterisk 1.4 Available for TESTING
might be old now hehe :) pre10 for libunicall, unicall, libsupertone, libmfcr2 0.0.3 for spandsp (final) and 0.0.4pre they were already out from testing folder/branch [EMAIL PROTECTED] wrote: Hi Nivlekch, Thanks for that, just a comment: What do you mean by new packages? new for spandsp, libmfcr2, unicall? chan_unicall? On 4/12/07, nivlekch [EMAIL PROTECTED] wrote: moises, guys, just an update, steve released new packages early april. i just did a successful compile, tomorrow i will test with a live e1 line. i managed to compile it with asterisk-1.4.2 a series of patches is on the way after a successful test. [EMAIL PROTECTED] wrote: nivlekch, nice to hear that :) I hope more people can test this. On 3/14/07, nivlekch [EMAIL PROTECTED] wrote: nice job moises, the hardwork you and steve put into chan_unicall is remarkable. with a little editing and tweaking, i was able to make the port to 1.4 here in the philippines without any problems. some part of libmfcr2 has to be changed for proper/better ANI exchage with PLDT(telco). looking good so far, better than the experience in 1.2, i'll post any update soon. anybody interfacing with PLDT interested, email me offline. [EMAIL PROTECTED] wrote: Im glad to let you know that finally I invested some time to make work Unicall in Asterisk 1.4, I must say not much testing could be done since I have no hardware available ( cards, servers ), however a friend was able to test it with a couple of calls with success, I need you to test this and report some feedback. The sources are available in: http://moy.ivsol.net/unicall/soft-switch/r1b1/ Kind Regards Moises Silva ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] polycom random reboots
On Mon, Apr 16, 2007 at 12:25:55PM +0200, Bas van der Veen wrote: Hello, Did you find anything while testing the LAN? Also, can you confirm that switching the switch, cabling, etc. did NOT solve the problem? It did not. We finally changed the server itself and reinstalled from a known-working installation at another of our sites. We also removed a 4BRI card percieved to be flaky (not needed on this 100% voip site). No more reboots since. I have spontaneous reboots with IP600's. ___ --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] duration sec and billing sec in cdr
Hi guys, i've installed asterisk to handle multiple voip accounts. I've looked at CDR configs, and managed to have cdr-csv files growing after each call. It would be easier to check my locak asterisk cdr's than logging into each account and check them at the provider website. i found that if i ring my sip softphone from my ata, bill seconds are counted correctly. however, if i call via a voip provider, bill seconds are counted incorrectly. Example: this call went to a pstn number New call from 551 --- 94361abcdefg (context: internal) Dialed: SIP/[EMAIL PROTECTED] Call start: 2007-04-14 20:10:55 Answered : 2007-04-14 20:10:55 Call end : 2007-04-14 20:11:10 Duration : 15 sec Bill : 15 sec this call went to my ata from the sip softphone: New call from 551 --- 505 (context: internal) Dialed: SIP/505|45 Call start: 2007-04-15 07:58:11 Answered : 2007-04-15 07:58:15 Call end : 2007-04-15 07:58:43 Duration : 32 sec Bill : 28 sec i've searched and google'd the wiki, but found only accounting software and cdr extensions for providers, but that's not what i need. thanks for any help Adam ___ --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] Instability on Asterisk
Oquendo, My provieder require sip digest authebtication: Asterisk send register to sip provider sip provider response with 401 asterisk send register again with authentication header sip provider response ok This is normal process, when problem happen, this process ocour until 401 message, and asterisk didn't add auth header. I thins this is a problem in asterisk becouse my ip phones can't register into him. After few minutes asterisk can register again and ip phones too. Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, J. Oquendo [EMAIL PROTECTED]: Frederico Madeira wrote: Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:19] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) Sniffing netowork i saw register packets arriving from ip phones, but asterisk didn't send response to it, and for external registry i saw register sended to sip provider, 401 response from sip provider and asterisk didn't start sip digest challenger, it was send a register message again without authentication header. Network connectivity for asterisk was ok during this problem moments. My asterisk is 1.4.2 with FC6 What can be wrong ? Thanks. Why do you believe this to be an issue with Asterisk. What you describe is this in barebones YourNetworkPhones --- connect --- SIP Provider SIP Provider --- starts handshake --- Yournetwork YourNetwork --- gets ball rolling --- SIP Provider SIP Provider ... ignores you Sounds like you should be ripping into your SIP provider they're sending you unauthorized messages which sounds like either they changed something, or you did. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] queue report problem
An open source queue log analyzer that we find useful... http://www.micpc.com/qloganalyzer/ in combination with CDR analysis... http://www.areski.net/asterisk-stat-v2/about.php regards, Drew Rilawich Ango wrote: HI all, I have a queue say 5000 and there are 10 member in the queue. When there is a call to the queue, the members will ring according to the defined strategy. In day end, I have to create a report about the queue and its member. But I found that it is very difficult to find the relation for the call to queue and the member who pick the call in CDR. Say, caller A calls the queue, queue member 9 pick the call. I want to know the caller A waiting time, conversion time for Caller A and member 9. Such relationship is very difficult to find in CDR. Anyone have such experience and how can I get such information? ango -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] FreePBX - Vicidial Integration
Hello, It's good to hear that you have success with VICIDIAL and FreeePBX, would you be willing to do any documentation on the steps you took to get this working reliably? Even something minimal on the VICIDIAL Wiki would be very helpful to a lot of people. http://eflo.net/VICIDIALwiki Thanks, MATT--- On 4/16/07, Erwan DESVERGNES [EMAIL PROTECTED] wrote: Hello, I've got actually near 10 Call Centers which works fine with FreePbx and Vicidial. Its right that if you use the FreePbx Dial Plan with macro it's very slow but you can use all freePbx stuff to create and manage Extension and Standard Pabx functions; and for vicidial you can create other dial plan with minimal things. -Message d'origine- De: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] De la part de Matt Florell Envoyé: vendredi 13 avril 2007 18:13 À: Asterisk Users Mailing List - Non-Commercial Discussion Objet: Re: [asterisk-users] FreePBX - Vicidial Integration On 4/13/07, Diego Quintana Cruz [EMAIL PROTECTED] wrote: Hi all, I am trying to install Vicidial in an existent FreePBX installation (I'm using Xorcom packages for Debian Etch), but I didn't find any documentation, I found only this guide [0], but is for trixbox only, do you think it will work on FreePBX on Etch? [0] http://iptn.org/vicidial/index.html Regards, Diego Quintana Cruz Hello, I would not recommend using FreePBX with VICIDIAL, mostly for efficiency and ease-of-use issues. The FreePBX calling path can contain dozens of steps, all slowing down and causing problems for VICIDIAL calls that are trying to go out. Not to mention the CallerID control issues that will cause you problems with a stock FreePBX/VICIDIAL system I usually recommend getting a separate server that goes to your FreePBX server over IAX if you will be using it in production. The VICIDIAL server would only have the sample VICIDIAL conf files and the changes needed to get your IAX trunk working. MATT--- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggishkeys - new info
On Apr 12, 2007, at 1:49 PM, Kevin P. Fleming wrote: Got off the phone with Polycom on this I have the same problem with my new 601 phone here (haven't seen the problem on the 650). I am using an IP650 with the latest firmware (and the corresponding sip.cfg file) and I have seen this behavior. It is most noticeable when on-hook dialing, where I will dial two or three digits and then press the fourth digit and nothing appears on the display for 1-2 seconds for that keypress. I am using new files with my 601/sidecar I have the issue and agree with Kevin, though I do mostly use it on hook. I have also noticed the end call or speakerphone button to be inoperative at times. It definately appears they have a bug and are not reading keypress in a timely fashion. I have also notice the sidecar has resumed its frequent rebooting again, had died down somewhat with the 2.0 code stream, but is back more often now with the 2.1. The phone is fine, but the sidecar will reboot randomly - whether idle or on a call. ___ --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] G.729 Pass-Thru Voicemail
Hello, I have just updated my Asterisk installation from 1.2x to 1.4 (on FreeBSD) - mostly everything seem to work fine. However, I use G.729 pass-thru - and I have before successfully used the following setup: http://www.voip-info.org/wiki/index.php?page=Asterisk%20G.729%20pass-thru However, it is not working with 1.4 - I see the following errors: [Apr 16 15:59:24] WARNING[10139]: channel.c:2816 set_format: Unable to find a codec translation path from g729 to gsm [Apr 16 15:59:24] WARNING[10139]: file.c:804 ast_streamfile: Unable to open vm-login (format 0x100 (g729)): No such file or directory [Apr 16 15:59:24] WARNING[10139]: app_voicemail.c:6104 vm_authenticate: Couldn't stream login file Is there any way I can get the voicemail functionality back, without reverting to gsm? Also - anyone know when native G.729 codec will be available for 1.4 on FreeBSD? Many thanks, Mike ___ --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] duration sec and billing sec in cdr
Sounds like your VoIP provider is incorrectly sending you an Answer before the call actually completes. I would contact your VoIP provider. I suppose it could also be possible that YOU have an Answer() statement that is only on your VoIP trunk. I would double check that, and then contact your VoIP provider to see if they have any suggestions. Basically, SOMEONE (your or voipstunt) is answering the call before it should be answered. On 4/16/07, Adam KOSA [EMAIL PROTECTED] wrote: Hi guys, i've installed asterisk to handle multiple voip accounts. I've looked at CDR configs, and managed to have cdr-csv files growing after each call. It would be easier to check my locak asterisk cdr's than logging into each account and check them at the provider website. i found that if i ring my sip softphone from my ata, bill seconds are counted correctly. however, if i call via a voip provider, bill seconds are counted incorrectly. Example: this call went to a pstn number New call from 551 --- 94361abcdefg (context: internal) Dialed: SIP/[EMAIL PROTECTED] Call start: 2007-04-14 20:10:55 Answered : 2007-04-14 20:10:55 Call end : 2007-04-14 20:11:10 Duration : 15 sec Bill : 15 sec this call went to my ata from the sip softphone: New call from 551 --- 505 (context: internal) Dialed: SIP/505|45 Call start: 2007-04-15 07:58:11 Answered : 2007-04-15 07:58:15 Call end : 2007-04-15 07:58:43 Duration : 32 sec Bill : 28 sec i've searched and google'd the wiki, but found only accounting software and cdr extensions for providers, but that's not what i need. thanks for any help Adam ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Loudspeaker
Hmm - just received an email from these guys last week. I know nothing about them. On Apr 15, 2007, at 9:23 PM, cb wrote: On Apr 15, 2007, at 9:53 PM, Klaverstyn, David C wrote: When a call comes in I want to ring an extension that happens to be loud speaker. The users can the press *8 to answer the call. Is there a SIP device that I can connect to Asterisk as an extension that can accomplish something like this? Do you already have the loud speaker? If not, I know there are various vendors of extension phone bells that do nothing more than plug into an analog line and ring the nice loud bell when a ring signal is received. You could easily combine one of those with a cheap ATA with FXS port. -chris www.mythtech.net ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Instability on Asterisk
While I don't use 1.4, it could be that the registration failure (you said 100 registration lines with your provider?!?) are blocking the phones from registering. This is only a guess, I don't know for sure. On 4/16/07, Frederico Madeira [EMAIL PROTECTED] wrote: Oquendo, My provieder require sip digest authebtication: Asterisk send register to sip provider sip provider response with 401 asterisk send register again with authentication header sip provider response ok This is normal process, when problem happen, this process ocour until 401 message, and asterisk didn't add auth header. I thins this is a problem in asterisk becouse my ip phones can't register into him. After few minutes asterisk can register again and ip phones too. Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, J. Oquendo [EMAIL PROTECTED]: Frederico Madeira wrote: Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:19] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) Sniffing netowork i saw register packets arriving from ip phones, but asterisk didn't send response to it, and for external registry i saw register sended to sip provider, 401 response from sip provider and asterisk didn't start sip digest challenger, it was send a register message again without authentication header. Network connectivity for asterisk was ok during this problem moments. My asterisk is 1.4.2 with FC6 What can be wrong ? Thanks. Why do you believe this to be an issue with Asterisk. What you describe is this in barebones YourNetworkPhones --- connect --- SIP Provider SIP Provider --- starts handshake --- Yournetwork YourNetwork --- gets ball rolling --- SIP Provider SIP Provider ... ignores you Sounds like you should be ripping into your SIP provider they're sending you unauthorized messages which sounds like either they changed something, or you did. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Need some dialplan help for obscure user request
I have a customer who wants their receptionist to input the users' long distance PINs for the because they use each others pins. I am having trouble coming up with a way to do this because of creating a channel between the user and receptionist, dropping the channel and its variables and creating a new one for the actual long distance call. Any advice is really needed. 1. User Dials Long Distance Number pattern --- calls receptionist's telephone 2. User tells receptionst that he/she needs to make a long distance call 3. Receptionist inputs User's pin from her phone (User doesn't get to know his pin) 4. Receptionist hangs up and long distance call is dialed using pin as the account code. Problems: 1. User Dials Long Distance Number pattern --- calls receptionist's telephone --- none, that's straight forward 2. User tells receptionst that he/she needs to make a long distance call --- none, that's straight forward 3. Receptionist inputs User's pin from her phone (User doesn't get to know his pin) How can I pick off receptionist's dtmf digits in the middle of the conversation? How can I assign those digits to salesman's account code, not receptionists? 4. Receptionist hangs up and long distance call is dialed using pin as the account code. I lose the channel variables at this point, how can I store salesman's PIN number do that it is available when the actual long distance number is dialed? ___ --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] duration sec and billing sec in cdr
Looks okay to me. either the number you are testing with your VoIP provider has an automated response which answers the call at the same sec you sent the Invite request or the provider is sending False Answer Supervision...do a sip debug and check while you make the call. On 4/16/07, Adam KOSA [EMAIL PROTECTED] wrote: Hi guys, i've installed asterisk to handle multiple voip accounts. I've looked at CDR configs, and managed to have cdr-csv files growing after each call. It would be easier to check my locak asterisk cdr's than logging into each account and check them at the provider website. i found that if i ring my sip softphone from my ata, bill seconds are counted correctly. however, if i call via a voip provider, bill seconds are counted incorrectly. Example: this call went to a pstn number New call from 551 --- 94361abcdefg (context: internal) Dialed: SIP/[EMAIL PROTECTED] Call start: 2007-04-14 20:10:55 Answered : 2007-04-14 20:10:55 Call end : 2007-04-14 20:11:10 Duration : 15 sec Bill : 15 sec this call went to my ata from the sip softphone: New call from 551 --- 505 (context: internal) Dialed: SIP/505|45 Call start: 2007-04-15 07:58:11 Answered : 2007-04-15 07:58:15 Call end : 2007-04-15 07:58:43 Duration : 32 sec Bill : 28 sec i've searched and google'd the wiki, but found only accounting software and cdr extensions for providers, but that's not what i need. thanks for any help Adam ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue trouble
Suity Zsolt wrote: Hi everyone, I'm in trouble with queue. There are a little local radio station with one studio and we have to switch queued callers to the live program. Everything works fine (counting callers, periodic announcements), but while the announcement is played for 'firs in line' caller, studio gets a free line out not the caller. Is the phone ringing when they pick up? If it's not ringing, the call is not being distributed to your agent. -Stephen- ___ --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] moving from asterisk1.2 to asterisk1.4
Hello all, I'm moving from Asterisk 1.2 to Asterisk 1.4.0-beta3. I'm just configuring an extension associated to an external sip-based voip service provider in order to be able to initiate/rcv pstn calls. Is there any relevant issue when moving from v1.2 to v1.4? Maybe something related to sip.conf/type variable? I'm only configuring sip.conf and extensions.conf config files. Thanks you very much, Victor ___ --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] G723 problems with TC400B
Jovanny Saravia wrote: Hello asteriskers, I hope someone could help me ... !! I bought a TC400B, and I am testing doing calls with G729 and G723. When I used G729 it works fine, but when I try to use G723 the RTP has very low quality and is not possible to hear to the other person in the phone. I try to find out the problem but the only weird that I saw in the debug info is this line (using asterisk 1.2.17): asterisk*CLI Apr 15 19:42:44 WARNING[17321] chan_zap.c: Frame too large Apr 15 19:42:48 WARNING[17321]: chan_zap.c:4931 zt_write: Frame too large Looks like you are going to have to modify the frame size for G723 on the phones. Not sure what frame size asterisk is expecting but I would guess 20ms or 30ms. This is my scenario: - 1 Intel(R) Core(TM)2 CPU (Core 2 Duo with 2.13GHz) - 1 Gigabit Memory - kernel 2.6.18-1.2798.fc6 x86_64 - OS: Fedora Core 6 I tried first with asterisk 1.4.1 and its dependencies but the problem is there, too much noise when someone speeaks (poor voice quality). In asterisk 1.2.17 and its dependencies (libpri, zaptel and addons) is a little better but the voice quality is not good anyway. In this scenario appears the Warning : Frame too large. The show transcoders and show translations looks fine in asterisk CLI: asterisk*CLI show transcoder 0/0 encoders/decoders of 92 channels (G.729a / G.723.1 5.3 kbps) are in use. asterisk*CLI show translation Translation times between formats (in milliseconds) Source Format (Rows) Destination Format(Columns) g723 gsm ulaw alaw g726 adpcm slin lpc10 g729 speex ilbc g723 - 4 1 1 3 3 2 5 2 -14 gsm 3 - 2 2 2 2 1 4 3 -13 ulaw 1 3 - 1 2 2 1 4 1 -13 alaw 1 3 1 - 2 2 1 4 1 -13 g726 3 3 2 2 - 2 1 4 3 -13 adpcm 3 3 2 2 2 - 1 4 3 -13 slin 2 2 1 1 1 1 - 3 2 -12 lpc10 4 4 3 3 3 3 2 - 4 -14 g729 2 4 1 1 3 3 2 5 - -14 speex - - - - - - - - - - - ilbc 4 4 3 3 3 3 2 5 4 - - asterisk*CLI After of this I upgrade the kernel to 2.6.20-1.2944.fc6 x86_64, and the problem remains. Any help will be so much appreciated. -- Jovanny Saravia Solutions Manager e-solutions Ltda [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] +57-310-7676163 ___ --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 -- Andres Technical Support http://www.telesip.net ___ --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] Voicemail: How to send a notification even if Caller does not let any messages?
Hi, First, sorry to repost, As I didn't get any replies, maybe this time, I will get more lucky. I was wondering if there was a way in Asterisk (agi script, asterisk-itself, whatever ... ) to send a notification to the user (Mail, SMS like voicemail application is doing) if the user has called, but did not leave any messages? I tried to use the minmessage, but, couldn't. Is that the way ? I was thinking of using the h Dialplan, and launch some script, but then, how to know if caller has left a message or not ? I wouldn't like to send 2 messages to the user. Thanks for your help ! Jean-Marc ___ --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] Fax with Asterisk + Hylafax
On Sun, 15 Apr 2007 22:10:34 -0700, Yuan LIU wrote From: Steve Totaro [EMAIL PROTECTED] Date: Sun, 15 Apr 2007 22:36:15 -0400 Stephen Bosch wrote: Steve Totaro wrote: You could try to get it working but it may never be 100%. If your needs are 100% then I suggest using a standard fax and get an analog line and do it the old fashioned way. If you need Hylafax type features then buy a modem that is compatible with Hylafax and run it on a different box. It's not entirely clear to me why people continue to cling to the idea that Asterisk should handle faxing also. What's the benefit? Hylafax is great, and you can even use it on the same machine. On same machine is a bit exaggerated, considering there is a Zaptel card on it. (But if Zaptel and Hylafax can share an X100P driver ...) You can have Asterisk and Hylafax on the same machine when you use IAXmodem. This is the way I fax in my office with 99% success rate. I am using a TDM04B card for my lines. I use a software called Avantfax that gives me an interface to send and receive faxes through a web page. I have implemented this solution with several clients and have up to 15 virtual IAXmodem + Hylafax sessions running. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 ___ --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] duration sec and billing sec in cdr
Hi, and thanks for the suggestions!
Matt wrote:
Sounds like your VoIP provider is incorrectly sending you an Answer before
the call actually completes. I would contact your VoIP provider.
I suppose it could also be possible that YOU have an Answer() statement
that
is only on your VoIP trunk. I would double check that, and then contact
your VoIP provider to see if they have any suggestions.
this is what's most likely as i have no experience in asterisk configs.
I've checked the extension.conf settins, they are:
exten = _94./_5[05][15],1,Playback(please_wait)
exten = _94./_5[05][15],n,Set(CALLERID(name)=my_voip_username)
exten = _94./_5[05][15],n,Dial(SIP/00${EXTEN:[EMAIL PROTECTED])
and for the internal numbers:
exten = _NXZ,1,Set(TIMEOUT(digit)=2)
exten = _NXZ,2,Dial(SIP/${EXTEN},45)
exten = _NXZ,3,VoiceMail(b${EXTEN})
exten = _NXZ,103,VoiceMail(u${EXTEN})
Basically, SOMEONE (your or voipstunt) is answering the call before it
should be answered.
i will check this with more voip providers to see if they or i have
messed up something (but it's probably going to be me, i just don't know
where to start looking).
thanks again
Adam
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Re: [asterisk-users] Need some dialplan help for obscure user request
What's wrong with: User calls receptionist gives her the number. Receptionist hits XFER on her phone... punches in pin and dials the number, then hits XFER to complete the transfer? This could all be done outside a dial-plan... just use the phone's transfer feature. If you MUST have asterisk do it, then use the features.conf to setup blind transfer. The receptionist can just blind transfer the extention to the external number. On a totally seperate note, you've got some other major issues if you can't trust your employees to keep their LD pins to themseles! On 4/16/07, J French [EMAIL PROTECTED] wrote: I have a customer who wants their receptionist to input the users' long distance PINs for the because they use each others pins. I am having trouble coming up with a way to do this because of creating a channel between the user and receptionist, dropping the channel and its variables and creating a new one for the actual long distance call. Any advice is really needed. 1. User Dials Long Distance Number pattern --- calls receptionist's telephone 2. User tells receptionst that he/she needs to make a long distance call 3. Receptionist inputs User's pin from her phone (User doesn't get to know his pin) 4. Receptionist hangs up and long distance call is dialed using pin as the account code. Problems: 1. User Dials Long Distance Number pattern --- calls receptionist's telephone --- none, that's straight forward 2. User tells receptionst that he/she needs to make a long distance call --- none, that's straight forward 3. Receptionist inputs User's pin from her phone (User doesn't get to know his pin) How can I pick off receptionist's dtmf digits in the middle of the conversation? How can I assign those digits to salesman's account code, not receptionists? 4. Receptionist hangs up and long distance call is dialed using pin as the account code. I lose the channel variables at this point, how can I store salesman's PIN number do that it is available when the actual long distance number is dialed? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Queue trouble
Correct, if I am understanding you correctly when an announcement is playing to the caller the caller is in 'limbo' until the announcement completes. Once the announcement completes, the caller will go through. On 4/16/07, Suity Zsolt [EMAIL PROTECTED] wrote: Hi everyone, I'm in trouble with queue. There are a little local radio station with one studio and we have to switch queued callers to the live program. Everything works fine (counting callers, periodic announcements), but while the announcement is played for 'firs in line' caller, studio gets a free line out not the caller. member = SIP/suich ;only 1 member strategy = ringall Asterisk 1.2.17 built by root Any idea what can I do with that? -- Suich ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail: How to send a notification even if Caller does not let any messages?
If the voicemail portion is reached, but hungup on, the extapp portion of the config file is still executed. So you could have an external app which does any number of things (IM, etc). Rob Jean-Marc Salsa wrote: Hi, First, sorry to repost, As I didn't get any replies, maybe this time, I will get more lucky. I was wondering if there was a way in Asterisk (agi script, asterisk-itself, whatever ... ) to send a notification to the user (Mail, SMS like voicemail application is doing) if the user has called, but did not leave any messages ? I tried to use the minmessage, but, couldn't. Is that the way ? I was thinking of using the h Dialplan, and launch some script, but then, how to know if caller has left a message or not ? I wouldn't like to send 2 messages to the user. Thanks for your help ! Jean-Marc ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Polycom 501 issue withlatest firmware: sluggishkeys - new info
I just had this issue, and fixed it with the 501Presumably the 601 has the same thing. If you had the old firmware before, and you forced your phone to re-register every x seconds, take that line out. The phone will become more responsive. To handle NAT, use nat.keepalive.epxires instead (new in 2.x) Let me know if that worked. Mike -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones Sent: Monday, April 16, 2007 09:58 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Polycom 501 issue withlatest firmware: sluggishkeys - new info On Apr 12, 2007, at 1:49 PM, Kevin P. Fleming wrote: Got off the phone with Polycom on this I have the same problem with my new 601 phone here (haven't seen the problem on the 650). I am using an IP650 with the latest firmware (and the corresponding sip.cfg file) and I have seen this behavior. It is most noticeable when on-hook dialing, where I will dial two or three digits and then press the fourth digit and nothing appears on the display for 1-2 seconds for that keypress. I am using new files with my 601/sidecar I have the issue and agree with Kevin, though I do mostly use it on hook. I have also noticed the end call or speakerphone button to be inoperative at times. It definately appears they have a bug and are not reading keypress in a timely fashion. I have also notice the sidecar has resumed its frequent rebooting again, had died down somewhat with the 2.0 code stream, but is back more often now with the 2.1. The phone is fine, but the sidecar will reboot randomly - whether idle or on a call. ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration sec and billing sec in cdr
Playback automatically answers the call unless you tell it not to. See:
show application playback in the Asterisk CLI.
Adam KOSA wrote:
Hi, and thanks for the suggestions!
Matt wrote:
Sounds like your VoIP provider is incorrectly sending you an Answer
before
the call actually completes. I would contact your VoIP provider.
I suppose it could also be possible that YOU have an Answer()
statement that
is only on your VoIP trunk. I would double check that, and then contact
your VoIP provider to see if they have any suggestions.
this is what's most likely as i have no experience in asterisk configs.
I've checked the extension.conf settins, they are:
exten = _94./_5[05][15],1,Playback(please_wait)
exten = _94./_5[05][15],n,Set(CALLERID(name)=my_voip_username)
exten = _94./_5[05][15],n,Dial(SIP/00${EXTEN:[EMAIL PROTECTED])
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Re: [asterisk-users] Re: New T1 Asterisk installation
On Mon, 16 Apr 2007, David Cook wrote: Remember, you don't need to activate all 23 lines so if you just need 15 then you can activate only that number. You also can have potentially hundreds of numbers that terminate on this group of lines. This makes some of your coding a little different because you no longer make an association between which port(s) ring and what number the caller dialed to get here. This is called DNIS (Dialed Number Identification System) (people don't flame me for the ANI/DNIS thing OK? Not relevant for this discussion). Not a flame :) I think you are referring to DNIS vs DNID. DNIS (Dialed Number Identification Service) is the number the caller dialed. ANI (Automatic Number Identification) is the number the caller called from. PRI can also deliver ANI2 (aka Info Digits) which can (if the ANI provider configured it) tell you what type of service the ANI came from -- hospital, hotel, prison, cell, etc. I've never gotten ANI2 working, so if you have, please enlighten me :) IMNSHO, PRI beats the [EMAIL PROTECTED] out of any other T1 flavor. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ --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] duration sec and billing sec in cdr
The playback wait command may be what's doing it.
On 4/16/07, Adam KOSA [EMAIL PROTECTED] wrote:
Hi, and thanks for the suggestions!
Matt wrote:
Sounds like your VoIP provider is incorrectly sending you an Answer
before
the call actually completes. I would contact your VoIP provider.
I suppose it could also be possible that YOU have an Answer() statement
that
is only on your VoIP trunk. I would double check that, and then contact
your VoIP provider to see if they have any suggestions.
this is what's most likely as i have no experience in asterisk configs.
I've checked the extension.conf settins, they are:
exten = _94./_5[05][15],1,Playback(please_wait)
exten = _94./_5[05][15],n,Set(CALLERID(name)=my_voip_username)
exten = _94./_5[05][15],n,Dial(SIP/00${EXTEN:[EMAIL PROTECTED])
and for the internal numbers:
exten = _NXZ,1,Set(TIMEOUT(digit)=2)
exten = _NXZ,2,Dial(SIP/${EXTEN},45)
exten = _NXZ,3,VoiceMail(b${EXTEN})
exten = _NXZ,103,VoiceMail(u${EXTEN})
Basically, SOMEONE (your or voipstunt) is answering the call before it
should be answered.
i will check this with more voip providers to see if they or i have
messed up something (but it's probably going to be me, i just don't know
where to start looking).
thanks again
Adam
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Re: [asterisk-users] Fax with Asterisk + Hylafax
Carlos Chavez wrote: You can have Asterisk and Hylafax on the same machine when you use IAXmodem. This is the way I fax in my office with 99% success rate. I am I've just compiled stats for the last 30 days on our system for management, info below: Failure Rate (0.26%) CallsCalls CallsTotalPercentage Successful Dropped Failed CallsCalls Dropped - Total1944 311 5 2260 16.00% NOTE: Calls Successful do not reflect the actual number of pages received Calls Failed that were successful on a second or third attempt are not listed Calls Dropped(Spammers) may be of the same source trying to make multiple attempts of sending Doug -- Ben Franklin quote: Those who would give up Essential Liberty to purchase a little Temporary Safety, deserve neither Liberty nor Safety. ___ --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] openvz resources
Awesome, any chance you can share your resource specs? Thanks Miles Asterisk works great with openvz. Ive run 4 VE's with combined average around 32 simultaneous calls at any time and you wouldn't know the difference. ___ --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] duration sec and billing sec in cdr
The Playback command is auto-answering the call. you can use
Playback(please_wait,noanswer) to fix it.
Joss.
On 4/16/07, Adam KOSA [EMAIL PROTECTED] wrote:
Hi, and thanks for the suggestions!
Matt wrote:
Sounds like your VoIP provider is incorrectly sending you an Answer
before
the call actually completes. I would contact your VoIP provider.
I suppose it could also be possible that YOU have an Answer() statement
that
is only on your VoIP trunk. I would double check that, and then contact
your VoIP provider to see if they have any suggestions.
this is what's most likely as i have no experience in asterisk configs.
I've checked the extension.conf settins, they are:
exten = _94./_5[05][15],1,Playback(please_wait)
exten = _94./_5[05][15],n,Set(CALLERID(name)=my_voip_username)
exten = _94./_5[05][15],n,Dial(SIP/00${EXTEN:[EMAIL PROTECTED])
and for the internal numbers:
exten = _NXZ,1,Set(TIMEOUT(digit)=2)
exten = _NXZ,2,Dial(SIP/${EXTEN},45)
exten = _NXZ,3,VoiceMail(b${EXTEN})
exten = _NXZ,103,VoiceMail(u${EXTEN})
Basically, SOMEONE (your or voipstunt) is answering the call before it
should be answered.
i will check this with more voip providers to see if they or i have
messed up something (but it's probably going to be me, i just don't know
where to start looking).
thanks again
Adam
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Re: [asterisk-users] Polycom 501 issue with latest firmware: sluggishkeys - new info
Hi All - Got off the phone with Polycom on this I have the same problem with my new 601 phone here (haven't seen the problem on the 650). I am using an IP650 with the latest firmware (and the corresponding sip.cfg file) and I have seen this behavior. It is most noticeable when on-hook dialing, where I will dial two or three digits and then press the fourth digit and nothing appears on the display for 1-2 seconds for that keypress. I am using new files with my 601/sidecar I have the issue and agree with Kevin, though I do mostly use it on hook. I have also noticed the end call or speakerphone button to be inoperative at times. It definately appears they have a bug and are not reading keypress in a timely fashion. I have also notice the sidecar has resumed its frequent rebooting again, had died down somewhat with the 2.0 code stream, but is back more often now with the 2.1. The phone is fine, but the sidecar will reboot randomly - whether idle or on a call. I've been running all versions of the firmware and haven't seen this at all, but maybe we can help ourselves and Polycom and try to narrow down the possible causes. The two possible causes that I can see 1) Some phones were part of a bad hardware run, or 2) the people who are seeing this problem are running a particular feature on the phone that is using buggy code. Is there a 3)? I'll wager on 2). If that's true, maybe we can narrow it down to the particular feature that's causing the problem. Aside from some custom Alert Info's and button remapping, my configuration is pretty stock. I haven't been using Presence, or buddy watching, or shared lines. Is there anyone else using any of these or any other special features that has this issue? Is there any commonality that we can establish? - Noah ___ --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] duration sec and billing sec in cdr
Adam KOSA wrote:
this is what's most likely as i have no experience in asterisk
configs. I've checked the extension.conf settins, they are:
exten = _94./_5[05][15],1,Playback(please_wait)
exten = _94./_5[05][15],n,Set(CALLERID(name)=my_voip_username)
exten = _94./_5[05][15],n,Dial(SIP/00${EXTEN:[EMAIL PROTECTED])
The Playback is your problem... you need to add |noanswer to the end of
that to prevent it from automatically answering the call before it plays
that recording.
Trevor
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Re: [asterisk-users] Call Recording Servers
Tom Lynn wrote: You could also look at Oreka at sourceforge. Tom, We are moving in that direction, but we don't have it in production yet. Since it is a packet sniffing solution, the limiting factor becomes the point at which the kernel starts to drop an unacceptable number of packets. A PF_RING www.ntop.org/PF_RING.html enabled version of libpcap can help to raise this point. There's some pretty sophisticated buffering going on, but it's still a good idea to dedicate a fast disk (or RAID) to writing the recordings. Keep in mind that Oreka aims to do one thing, while Asterisk is a sort of VOIP Swiss Army knife. If all you want to do is record calls, Oreka is a good candidate. On the other hand, Asterisk will give you call recording along with a plethora of other features. Oreka's main developer is extremely skilled, helpful, and responsive. As far as dimensioning Oreka, here is a quote from him: What I can say is that we do have a customer recording 200 concurrent conversations without drops under Linux FC4 with the following server (Desktop hardware actually): Dell Dimension 9200 IntelR CoreTM 2 Duo Processor E6300 (2MB L2 Cache,1.86GHz,1066) 1 Gig of RAM Dual-Channel DDR2 SDRAM (533MHz) 1 x 80 Gig SATA II drive 1 x 300 Gig SATA I drive And this is without PF_RING or anything else, so I would be surprised if we could not push this further. Good day, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer ___ --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] Voicemail: How to send a notification even if Caller does not let any messages?
Thanks for the answer, I already use this extapp, to set on another server the MWI. But how to know if user has not let a message ? One could guess that 0 is the number of message to trigger such a notification ... but 0 is the number of message as well when you erase all your messages, so you shouldn't send a notification in that case. Any idea please ? Thanks, JM On 4/16/07, Rob Schall [EMAIL PROTECTED] wrote: If the voicemail portion is reached, but hungup on, the extapp portion of the config file is still executed. So you could have an external app which does any number of things (IM, etc). Rob Jean-Marc Salsa wrote: Hi, First, sorry to repost, As I didn't get any replies, maybe this time, I will get more lucky. I was wondering if there was a way in Asterisk (agi script, asterisk-itself, whatever ... ) to send a notification to the user (Mail, SMS like voicemail application is doing) if the user has called, but did not leave any messages ? I tried to use the minmessage, but, couldn't. Is that the way ? I was thinking of using the h Dialplan, and launch some script, but then, how to know if caller has left a message or not ? I wouldn't like to send 2 messages to the user. Thanks for your help ! Jean-Marc -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] moving from asterisk1.2 to asterisk1.4
Hi Victor - I'm moving from Asterisk 1.2 to Asterisk 1.4.0-beta3. I'm just configuring an extension associated to an external sip-based voip service provider in order to be able to initiate/rcv pstn calls. Is there any relevant issue when moving from v1.2 to v1.4? Maybe something related to sip.conf/type variable? I'm only configuring sip.conf and extensions.conf config files. You really don't want to run 1.4.0-beta3. Is there any reason you would not run 1.4.2?There have been many, many bug fixes since 1.4.0-beta3. There is a file called UPGRADE.txt in the asterisk source code that lists relevant changes between 1.2.x and 1.4.x. You may want to read that. The sample sip.conf file also has a great deal of information. - Noah ___ --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] Asterisk-Java website
Well, it _was_ up again Friday, and now it's down again Monday! :( Moises Silva wrote: Hum, I know Stefan, he is an asterisk-java dev, but he is not online right now, I will let him know ASAP. Thanks! On 4/12/07, Doug Garstang [EMAIL PROTECTED] wrote: Does anyone know who maintains the Asterisk-java web site at asterisk-java.org? The site seems to have been unavailable for a couple of days now. Doug ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail: How to send a notification even if Caller does not let any messages?
And by the way, I forgot, If I remember carefully, there is not so much info passed to this script (VM Number, context Number of messages) ... So for example, how do you get the caller ID info ? Thanks again, JM On 4/16/07, Jean-Marc Salsa [EMAIL PROTECTED] wrote: Thanks for the answer, I already use this extapp, to set on another server the MWI. But how to know if user has not let a message ? One could guess that 0 is the number of message to trigger such a notification ... but 0 is the number of message as well when you erase all your messages, so you shouldn't send a notification in that case. Any idea please ? Thanks, JM On 4/16/07, Rob Schall [EMAIL PROTECTED] wrote: If the voicemail portion is reached, but hungup on, the extapp portion of the config file is still executed. So you could have an external app which does any number of things (IM, etc). Rob Jean-Marc Salsa wrote: Hi, First, sorry to repost, As I didn't get any replies, maybe this time, I will get more lucky. I was wondering if there was a way in Asterisk (agi script, asterisk-itself, whatever ... ) to send a notification to the user (Mail, SMS like voicemail application is doing) if the user has called, but did not leave any messages ? I tried to use the minmessage, but, couldn't. Is that the way ? I was thinking of using the h Dialplan, and launch some script, but then, how to know if caller has left a message or not ? I wouldn't like to send 2 messages to the user. Thanks for your help ! Jean-Marc -- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: New T1 Asterisk installation
Steve Edwards wrote: On Mon, 16 Apr 2007, David Cook wrote: Remember, you don't need to activate all 23 lines so if you just need 15 then you can activate only that number. You also can have potentially hundreds of numbers that terminate on this group of lines. This makes some of your coding a little different because you no longer make an association between which port(s) ring and what number the caller dialed to get here. This is called DNIS (Dialed Number Identification System) (people don't flame me for the ANI/DNIS thing OK? Not relevant for this discussion). Not a flame :) I think you are referring to DNIS vs DNID. DNIS (Dialed Number Identification Service) is the number the caller dialed. ANI (Automatic Number Identification) is the number the caller called from. PRI can also deliver ANI2 (aka Info Digits) which can (if the ANI provider configured it) tell you what type of service the ANI came from -- hospital, hotel, prison, cell, etc. I've never gotten ANI2 working, so if you have, please enlighten me :) IMNSHO, PRI beats the [EMAIL PROTECTED] out of any other T1 flavor. Thanks in advance, Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 I used to get ANI2 on some UCN PRIs. You could see the info in PRI Debug (a two digit code) but I am not sure where you can populate that data. I guess you just have to ask your provider for it, I inherited that setup. Thanks, Steve ___ --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] Voicemail: How to send a notification even if Caller does not let any messages?
There is no absolute way to verify if the user left a new message, since it only tells you how many messages are currently in the box. If not many messages are sent you could do a stat on the newest voicemail file in their new folder. Then see if its more than a few seconds old. If its only a few seconds old, then the message was just left. But like I said, still a tough thing to guarentee. Rob Jean-Marc Salsa wrote: Thanks for the answer, I already use this extapp, to set on another server the MWI. But how to know if user has not let a message ? One could guess that 0 is the number of message to trigger such a notification ... but 0 is the number of message as well when you erase all your messages, so you shouldn't send a notification in that case. Any idea please ? Thanks, JM On 4/16/07, *Rob Schall* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: If the voicemail portion is reached, but hungup on, the extapp portion of the config file is still executed. So you could have an external app which does any number of things (IM, etc). Rob Jean-Marc Salsa wrote: Hi, First, sorry to repost, As I didn't get any replies, maybe this time, I will get more lucky. I was wondering if there was a way in Asterisk (agi script, asterisk-itself, whatever ... ) to send a notification to the user (Mail, SMS like voicemail application is doing) if the user has called, but did not leave any messages ? I tried to use the minmessage, but, couldn't. Is that the way ? I was thinking of using the h Dialplan, and launch some script, but then, how to know if caller has left a message or not ? I wouldn't like to send 2 messages to the user. Thanks for your help ! Jean-Marc ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com http://Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration sec and billing sec in cdr
This is interesting to me.. I'm a newbie, so please forgive a dumb
question, but what use is it to play a message if you don't pick up the
phone first?? Who's hearing it?
-Original Message-
Adam KOSA wrote:
this is what's most likely as i have no experience in asterisk
configs. I've checked the extension.conf settins, they are:
exten = _94./_5[05][15],1,Playback(please_wait)
exten = _94./_5[05][15],n,Set(CALLERID(name)=my_voip_username)
exten = _94./_5[05][15],n,Dial(SIP/00${EXTEN:[EMAIL PROTECTED])
The Playback is your problem... you need to add |noanswer to the end of
that to prevent it from automatically answering the call before it plays
that recording.
Trevor
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Re: [asterisk-users] Passing a variable from one Asterisk box toanother
Hi Craig
I've been developing a Recording Server app (which I will be giving back
to the community) and one of the requirements is for the recording to be
offloaded to several machines. Because of the filename is being set prior to
the recording, I need to pass this variable to the slave server. I'm using
1.2.13 (heavily patched) and I came across your email. Any chance of getting
your port? Thanks for your help...
Jesus Mogollon
On 2/22/07, Craig Guy [EMAIL PROTECTED] wrote:
Hi Richard,
there was a thread regarding this a while ago on the dev list which
resulted
in a patch being made to allow variable passing via IAX2 channels. See
http://bugs.digium.com/view.php?id=7619 for the patch which I think is in
SVN or anyhow, is not in 1.2
I have recently backported this patch to 1.2 and have a patch which is
tested against 1.2.12, 1.2.12.1 and 1.2.15, but should work against at
least
1.2.13 and 1.2.14. The patch introduces a new dialplan function called
IAXVAR, Email me if interested.
Craig
- Original Message -
From: Richard Lyman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 21, 2007 7:27 AM
Subject: Re: [asterisk-users] Passing a variable from one Asterisk box
toanother
Richard Lyman wrote:
Eric Bishop wrote:
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All
works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call dials into Box 2 the variable Foo does not get passed...
Does anyone have any clever ideas?
as noted in asterisk/docs/README.variables (iirc)
you should see that variable inheritance can occur by prefacing the
variable with '_' or '__'
also, depending on the age of your asterisk you might want to start
using
'Set' vice 'SetVar'
also, having ${EXTEN:0} , the :0 doesn't do anything, so you should not
use it and just have ${EXTEN}
i hope this helps
sadly replying to my own post, but, i forgot to mention that
passing variables with IAX2 can be an issue sometimes when you use
user and peer (the user side can pass vars the peer side can not, or
doesn't accept them iirc)
this does not happen using friend, but that has its own issues... check
the wiki for more thoughts about this.
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[asterisk-users] jittershrinkrate equivalent in current (new) iax jb implementation
hello, is there any equivalent, that is currently usefull, if I have some iax connections with jitter spikes and another with minimal jitter? for my jittery connections, I don't like to shrink jitter buffer too fast, because another jitter spike can occur again and small jb can't cover it. as I read, in older iax jb implementation, this can be solved using jittershrinkrate= option, why it was currently removed? :-\ PJ ___ --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] BSNL caller ID (India)
Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Can someone please help. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] duration sec and billing sec in cdr
Steve Jones wrote: This is interesting to me.. I'm a newbie, so please forgive a dumb question, but what use is it to play a message if you don't pick up the phone first?? Who's hearing it? Many types of connections allow you to do early audio or on hook audio. A perfect example of this is when you call a disconnected number, you get the telco audio message, but don't get billed for the callbecause the telco never answered the line. ___ --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] Redundant * servers
Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ /\ User2 -- Go to register --- SIP2 - Whereis? -- DB / \/ User3 -- -- SIP3 -- Where users no matter who they are, register and are passed off to the next server in sequence... For example, ten people are all registering right now... User1 -- SIP1 User2 -- SIP2 User3 -- SIP3 User4 -- SIP1 And so on... where an ATA, VoIP phone, etc., would have its information stored via database and pulled and pushed anytime something happened with that User... Make sense? Think of a load balanced SIP cluster if you will WITHOUT SER or Dundi... -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams smime.p7s Description: S/MIME Cryptographic Signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: New T1 Asterisk installation
Steve Edwards wrote: On Mon, 16 Apr 2007, David Cook wrote: Remember, you don't need to activate all 23 lines so if you just need 15 then you can activate only that number. You also can have potentially hundreds of numbers that terminate on this group of lines. This makes some of your coding a little different because you no longer make an association between which port(s) ring and what number the caller dialed to get here. This is called DNIS (Dialed Number Identification System) (people don't flame me for the ANI/DNIS thing OK? Not relevant for this discussion). Not a flame :) I think you are referring to DNIS vs DNID. DNIS (Dialed Number Identification Service) is the number the caller dialed. ANI (Automatic Number Identification) is the number the caller called from. PRI can also deliver ANI2 (aka Info Digits) which can (if the ANI provider configured it) tell you what type of service the ANI came from -- hospital, hotel, prison, cell, etc. I've never gotten ANI2 working, so if you have, please enlighten me :) IMNSHO, PRI beats the [EMAIL PROTECTED] out of any other T1 flavor. (Is there another T1 flavour? I thought PRI was it :) BRI isn't T1 anymore) There are some parts of the world where you can't get partial PRI anymore, and there's an ugly unserved gap in between analog lines and a full PRI :( $650 minimum for a full PRI around here. You need 17 lines before that makes financial sense. -Stephen- ___ --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] Huh? IP address ending with 611
Looks like a PolycomSoundPointIP bug to me. The Via header, Contact both has 66.38.177.611:5060 Thanks, Neel -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- [EMAIL PROTECTED] On Behalf Of Mike Sent: Friday, April 13, 2007 7:28 AM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [asterisk-users] Huh? IP address ending with 611 The weird thing is that the phone actually works for now, but I want to proactively fix anything that may go wrong (this phone _has_ to work until Saturday) A SIP debug gives me this: --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms hd-t3143cl*CLI sip -- SIP read from 66.38.177.61:5060: REGISTER sip:pbx.test.ca SIP/2.0 Via: SIP/2.0/UDP 66.38.177.611:5060;branch=z9hG4bKcf7e2b60CAE36701 From: notarius-phone-1 sip:[EMAIL PROTECTED];tag=28A4D21C-80220BAB To: sip:[EMAIL PROTECTED] CSeq: 1803 REGISTER Call-ID: [EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED]:5060;methods=INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_501-UA/1.6.7.0098 Authorization: Digest username=notarius-phone-1, realm=asterisk, nonce=20d11a72, uri=sip:pbx.test.ca, response=dbcfab79977a81ea3681bbe574bd1c37, algorithm=MD5 Max-Forwards: 70 Expires: 30 Content-Length: 0 --- (12 headers 0 lines) --- Using latest REGISTER request as basis request Apr 13 08:28:26 WARNING[20348]: chan_sip.c:7036 check_via: '66.38.177.611' is not a valid host Transmitting (NAT) to 66.38.177.61:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 66.38.177.611:5060;branch=z9hG4bKcf7e2b60CAE36701;received=66.38.177.61 From: notarius-phone-1 sip:[EMAIL PROTECTED];tag=28A4D21C-80220BAB To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] Seq: 1803 REGISTERp User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- Transmitting (NAT) to 66.38.177.61:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 66.38.177.611:5060;branch=z9hG4bKcf7e2b60CAE36701;received=66.38.177.61 From: notarius-phone-1 sip:[EMAIL PROTECTED];tag=28A4D21C-80220BAB To: sip:[EMAIL PROTECTED];tag=as42c283c1 Call-ID: [EMAIL PROTECTED] CSeq: 1803 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Expires: 30 Contact: sip:[EMAIL PROTECTED]:5060;expires=30 Date: Fri, 13 Apr 2007 12:28:26 GMT Content-Length: 0 -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alex Balashov Sent: Thursday, April 12, 2007 23:20 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Huh? IP address ending with 611 Can you do a packet capture and see what the actual contact (Via) in fact says right before it hits Asterisk? On Thu, 12 Apr 2007, Mike said something to this effect: Hi, I`m getting this (from one of my registered phone that has been installed at some location I can`t access at the moment) in the Asterisk CLI. Notice the last 3 digits of the IP address in the error message: Apr 12 23:06:16 WARNING[25593]: chan_sip.c:7036 check_via: '67.39.117.611' is not a valid host Of course it's not a valid host! But, when using sip show peers, the phone is actually listed with IP address 67.39.117.61 (which makes alot more sense, but then again I shouldn`t be getting any warning in this case). Where is that error coming from then? Are there any consequences? Using 1.2.13. Mike -- Alex Balashov [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk-Java website
Doug Garstang wrote: Well, it _was_ up again Friday, and now it's down again Monday! :( sorry, there seem to be problem with the nameservers. I'll hava a look at it asap. =Stefan signature.asc Description: OpenPGP digital signature ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail - digits/1F does not exist in any format
I've seen this before, in an ISDN card (can't recall which one) that defaults the incoming language to german. Since you don't have german, it defaults to english files but voicemail still runs through the german logic (e.g. 1F for femail). I reported a bug against this, it was silently killing the call - no error handling. I suggested that they check if the desired language is installed and if not, that within the app the 'temporarily change' the language to english so that it doesn't go off looking for sound files that are not there. I can't recall the bug number - but they didn't feel it was a reasonable approach ... different opinions I guess, they decided the behavior was accetable. philippe From: Per Jessen [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Sun, 15 Apr 2007 14:55:31 +0200 Subject: Re: [asterisk-users] voicemail - digits/1F does not exist in any format Carlos Chavez wrote: I am assuming you are using a language other that English? If so, do you have the language files installed in the correct place? For asterisk 1.2 you need a structure like this: No, I'm using English. The default setup that came with 1.4.1. The other sound files are in /var/lib/asterisk/sounds/digits: -rw-rw-r-- 1 per 1000 1353 Feb 20 23:05 0.gsm -rw-rw-r-- 1 per 1000 1089 Feb 20 23:05 1.gsm -rw-rw-r-- 1 per 1000 1023 Feb 20 23:05 10.gsm -rw-rw-r-- 1 per 1000 1353 Feb 20 23:05 11.gsm -rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 12.gsm -rw-rw-r-- 1 per 1000 1485 Feb 20 23:05 13.gsm -rw-rw-r-- 1 per 1000 1485 Feb 20 23:05 14.gsm -rw-rw-r-- 1 per 1000 1518 Feb 20 23:05 15.gsm -rw-rw-r-- 1 per 1000 1617 Feb 20 23:05 16.gsm -rw-rw-r-- 1 per 1000 1782 Feb 20 23:05 17.gsm -rw-rw-r-- 1 per 1000 1551 Feb 20 23:05 18.gsm -rw-rw-r-- 1 per 1000 1650 Feb 20 23:05 19.gsm -rw-rw-r-- 1 per 1000 990 Feb 20 23:05 2.gsm -rw-rw-r-- 1 per 1000 1254 Feb 20 23:05 20.gsm -rw-rw-r-- 1 per 1000 990 Feb 20 23:05 3.gsm -rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 30.gsm -rw-rw-r-- 1 per 1000 1089 Feb 20 23:05 4.gsm -rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 40.gsm -rw-rw-r-- 1 per 1000 1122 Feb 20 23:05 5.gsm -rw-rw-r-- 1 per 1000 1419 Feb 20 23:05 50.gsm -rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 6.gsm -rw-rw-r-- 1 per 10000 Feb 20 23:05 60.gsm -rw-rw-r-- 1 per 1000 1320 Feb 20 23:05 7.gsm -rw-rw-r-- 1 per 1000 1485 Feb 20 23:05 70.gsm -rw-rw-r-- 1 per 1000 891 Feb 20 23:05 8.gsm -rw-rw-r-- 1 per 1000 1155 Feb 20 23:05 80.gsm -rw-rw-r-- 1 per 1000 1254 Feb 20 23:05 9.gsm -rw-rw-r-- 1 per 1000 1419 Feb 20 23:05 90.gsm /Per Jessen, Zürich - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] BSNL caller ID (India)
Hello Mr. Sanjay, I tried a lot to get caller ID in India. But, It doesn't work. I came to know that Its not possible to get caller ID in India (Not only in India, don't get caller ID in some countrys). Thank you. Regards, Chandra. Sanjay Rajdev [EMAIL PROTECTED] wrote: Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Can someone please help. Regards, Sanjay Rajdev ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users - Ahhh...imagining that irresistible new car smell? Check outnew cars at Yahoo! Autos.___ --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] Redundant * servers
Use round robin on DNS with a replicated DB on each server On 4/16/07, J. Oquendo [EMAIL PROTECTED] wrote: Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ /\ User2 -- Go to register --- SIP2 - Whereis? -- DB / \/ User3 -- -- SIP3 -- Where users no matter who they are, register and are passed off to the next server in sequence... For example, ten people are all registering right now... User1 -- SIP1 User2 -- SIP2 User3 -- SIP3 User4 -- SIP1 And so on... where an ATA, VoIP phone, etc., would have its information stored via database and pulled and pushed anytime something happened with that User... Make sense? Think of a load balanced SIP cluster if you will WITHOUT SER or Dundi... -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- --- Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) [EMAIL PROTECTED] - [EMAIL PROTECTED] If any of the above are down we have bigger problems than my email! Hind sight is most always 20/20 or better. --- ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] IAX implementation question
People, I've setup Asterisk in a basic mode with SIP protocol. In the future I wanna connect several offices each one with an own Asterisk server, using IAX because I read it has no firewalling problems using just one UDP port for control and data -aming other advantages- . SIP has NAT problems I know. Do you recommend the use of IAX instead of SIP for users and among several Asterisk's ??? Does the IAX implementation take any extra considerations than SIP ??? Any initial guide for IAX - Asterisk configuration ??? Thanks a lot, Alejandro ___ --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] duration sec and billing sec in cdr
Yossi Ben Hagai wrote: The Playback command is auto-answering the call. you can use Playback(please_wait,noanswer) to fix it. thanks a lot to everyone who answered, this, of course solved this issue, it's also in the doc, i just didn't have the idea to look at playback's manual :( regards adam ___ --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] Instability on Asterisk
Matt, It's make no sense. Asterisk should process messages in diferents threds, not in queue. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, Matt [EMAIL PROTECTED]: While I don't use 1.4, it could be that the registration failure (you said 100 registration lines with your provider?!?) are blocking the phones from registering. This is only a guess, I don't know for sure. On 4/16/07, Frederico Madeira [EMAIL PROTECTED] wrote: Oquendo, My provieder require sip digest authebtication: Asterisk send register to sip provider sip provider response with 401 asterisk send register again with authentication header sip provider response ok This is normal process, when problem happen, this process ocour until 401 message, and asterisk didn't add auth header. I thins this is a problem in asterisk becouse my ip phones can't register into him. After few minutes asterisk can register again and ip phones too. Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, J. Oquendo [EMAIL PROTECTED]: Frederico Madeira wrote: Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for ' [EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:19] NOTICE[2851] chan_sip.c:-- Registration for ' [EMAIL PROTECTED]' timed out, trying again (Attempt #36) Sniffing netowork i saw register packets arriving from ip phones, but asterisk didn't send response to it, and for external registry i saw register sended to sip provider, 401 response from sip provider and asterisk didn't start sip digest challenger, it was send a register message again without authentication header. Network connectivity for asterisk was ok during this problem moments. My asterisk is 1.4.2 with FC6 What can be wrong ? Thanks. Why do you believe this to be an issue with Asterisk. What you describe is this in barebones YourNetworkPhones --- connect --- SIP Provider SIP Provider --- starts handshake --- Yournetwork YourNetwork --- gets ball rolling --- SIP Provider SIP Provider ... ignores you Sounds like you should be ripping into your SIP provider they're sending you unauthorized messages which sounds like either they changed something, or you did. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Re: New T1 Asterisk installation
On Mon, 16 Apr 2007, Stephen Bosch said something to this effect: There are some parts of the world where you can't get partial PRI anymore, and there's an ugly unserved gap in between analog lines and a full PRI :( Even where you can get a fractional PRI, there's not really a lot of incentive for the LEC to substantially discount the circuit because the underlying physical layer, loop, etc. is still running at a T1 rate. The fact that you can turn off some of the channels doesn't inherently decrease their costs unless there's a huge correlate in decreased usage. $650 minimum for a full PRI around here. You need 17 lines before that makes financial sense. Who quoted you this? Where are you? This sounds like the extortion that BellSouth pulls down here. Do not order PRIs from the ILEC. They blow. If you contact me off-list I might be able to help you find some cheaper options. It is generally reasonable to expect to find a PRI for about half that price. The best way is to get someone who buys a lot of transport from a carrier to resell it to you because their costs on it are many times lower due to the fact that they buy in large volumes. For example, many dialup wholesalers with their own facilities order entire DS3s worth of the stuff, in which case they may be able to cut you a much better deal. -- Alex -- Alex Balashov [EMAIL PROTECTED] ___ --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] Redundant * servers
It's possible, have the SIP clients use SRV records for server location and use asterisk ARA to store SIP peers and extension.conf on DB. if the users are not behind NAT it should work. (open)SER is much better solution for high traffic / availability setups. On 4/16/07, J. Oquendo [EMAIL PROTECTED] wrote: Without using Dundi or SER, any thoughts on the following anyone? Has something similar been implemented anywhere so as to me not having to horribly butcher code... 4 servers SIP1-4 User1 -- -- SIP1 -- \ /\ User2 -- Go to register --- SIP2 - Whereis? -- DB / \/ User3 -- -- SIP3 -- Where users no matter who they are, register and are passed off to the next server in sequence... For example, ten people are all registering right now... User1 -- SIP1 User2 -- SIP2 User3 -- SIP3 User4 -- SIP1 And so on... where an ATA, VoIP phone, etc., would have its information stored via database and pulled and pushed anytime something happened with that User... Make sense? Think of a load balanced SIP cluster if you will WITHOUT SER or Dundi... -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] voicemail - digits/1F does not exist in any format
Philippe Lindheimer wrote: I've seen this before, in an ISDN card (can't recall which one) that defaults the incoming language to german. Since you don't have german, it defaults to english files but voicemail still runs through the german logic (e.g. 1F for femail). I reported a bug against this, it was silently killing the call - no error handling. I suggested that they check if the desired language is installed and if not, that within the app the 'temporarily change' the language to english so that it doesn't go off looking for sound files that are not there. I can't recall the bug number - but they didn't feel it was a reasonable approach ... different opinions I guess, they decided the behavior was accetable. philippe Odd. You'd think the developers would want SOME kind of exception handling with CLI output so that you'd have at least a guess at what the problem was. -- Warm Regards, Lee ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Stuck on MySQL UPDATE
What I'm retrying to do is update mysql field with the new message ID
that was just recorded. Ideally, I'd like to specify
the field to update using a variable ${BINID} and use ${NEWPHRASENAME}
for the value - I'm not sure asterisk will allow
using a variable for the field name and if not, I'll attempt to create
an exten for each bin to update.
Here the method I'd like to use:
exten = sav,n,MYSQL(Connect connid localhost root password dax)
exten = sav,n,MYSQL(QUERY resultid ${connid}UPDATE\ dnislookup\ SET\
${BINID}\ =\ ${NEWPHRASENAME}\ WHERE\ dnis\ =\ ${IVR-Exten})
But I've tried this too:
exten = sav,n,MYSQL(Connect connid localhost root password dax)
exten = sav,n,MYSQL(QUERY resultid ${connid}UPDATE\ dnislookup\ SET\
bin2\ =\ ${NEWPHRASENAME}\ WHERE\ dnis\ =\ ${IVR-Exten})
However, neither one of these saves to new value into the bin2 (or
${BINID}) field.
From the logs:
Apr 16 12:40:05 VERBOSE[13718] logger.c: == Where Field Name = bin2 and
value to update is 2_4643
Apr 16 12:40:05 DEBUG[13718] pbx.c: Launching 'MYSQL'
Apr 16 12:40:05 DEBUG[13718] pbx.c: Launching 'MYSQL'
Apr 16 12:40:05 WARNING[13718] app_addon_sql_mysql.c: aMYSQL_query:
missing some arguments
Apr 16 12:40:05 DEBUG[13718] pbx.c: Launching 'MYSQL'
Apr 16 12:40:05 WARNING[13718] app_addon_sql_mysql.c: Identifier 160,
identifier_type 2 not found in identifier list
Apr 16 12:40:05 WARNING[13718] app_addon_sql_mysql.c: Invalid result
identifier 160 passed in aMYSQL_clear
Apr 16 12:40:05 DEBUG[13718] pbx.c: Launching 'MYSQL'
Can you suggest something?
Bart
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Re: [asterisk-users] Instability on Asterisk
I understand... but I know, at least in 1.2 if there was a DNS failure for some reason asterisk stopped doing anything else. That is... if I restart asterisk and it goes to register with , say, my 6 SIP upstream peers... but they are timing out for some reason asterisk won't initialize zap, or other sip or IAX stuff until it times out all 6 of those. I was under the impression this was being fixed in 1.4, but maybe it has not been. On 4/16/07, Frederico Madeira [EMAIL PROTECTED] wrote: Matt, It's make no sense. Asterisk should process messages in diferents threds, not in queue. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, Matt [EMAIL PROTECTED]: While I don't use 1.4, it could be that the registration failure (you said 100 registration lines with your provider?!?) are blocking the phones from registering. This is only a guess, I don't know for sure. On 4/16/07, Frederico Madeira [EMAIL PROTECTED] wrote: Oquendo, My provieder require sip digest authebtication: Asterisk send register to sip provider sip provider response with 401 asterisk send register again with authentication header sip provider response ok This is normal process, when problem happen, this process ocour until 401 message, and asterisk didn't add auth header. I thins this is a problem in asterisk becouse my ip phones can't register into him. After few minutes asterisk can register again and ip phones too. Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, J. Oquendo [EMAIL PROTECTED]: Frederico Madeira wrote: Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for '[EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for ' [EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:19] NOTICE[2851] chan_sip.c:-- Registration for ' [EMAIL PROTECTED]' timed out, trying again (Attempt #36) Sniffing netowork i saw register packets arriving from ip phones, but asterisk didn't send response to it, and for external registry i saw register sended to sip provider, 401 response from sip provider and asterisk didn't start sip digest challenger, it was send a register message again without authentication header. Network connectivity for asterisk was ok during this problem moments. My asterisk is 1.4.2 with FC6 What can be wrong ? Thanks. Why do you believe this to be an issue with Asterisk. What you describe is this in barebones YourNetworkPhones --- connect --- SIP Provider SIP Provider --- starts handshake --- Yournetwork YourNetwork --- gets ball rolling --- SIP Provider SIP Provider ... ignores you Sounds like you should be ripping into your SIP provider they're sending you unauthorized messages which sounds like either they changed something, or you did. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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] ${CALLERIDNUM} DEPRECATED in 1.4.2
${CALLERIDNUM} is DEPRECATED in 1.4.2 what is the alternative to find the same
in extensions.conf for setting a proper dialplan.
Please Suggest
Regards,
Sanjay Rajdev
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Re: [asterisk-users] Passing a variable from one Asterisk boxtoanother
From: Jesus Mogollon [EMAIL PROTECTED]
Date: Mon, 16 Apr 2007 13:33:16 -0400
Hi Craig
I've been developing a Recording Server app (which I will be giving back
to the community) and one of the requirements is for the recording to be
offloaded to several machines. Because of the filename is being set prior
to
the recording, I need to pass this variable to the slave server. I'm using
1.2.13 (heavily patched) and I came across your email. Any chance of
getting
your port? Thanks for your help...
If there are only a limited number of variables to pass, you may as well do
this in dial plan using SIPHEADER.
Yuan Liu
Jesus Mogollon
On 2/22/07, Craig Guy [EMAIL PROTECTED] wrote:
Hi Richard,
there was a thread regarding this a while ago on the dev list which
resulted
in a patch being made to allow variable passing via IAX2 channels. See
http://bugs.digium.com/view.php?id=7619 for the patch which I think is in
SVN or anyhow, is not in 1.2
I have recently backported this patch to 1.2 and have a patch which is
tested against 1.2.12, 1.2.12.1 and 1.2.15, but should work against at
least
1.2.13 and 1.2.14. The patch introduces a new dialplan function called
IAXVAR, Email me if interested.
Craig
- Original Message -
From: Richard Lyman [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, February 21, 2007 7:27 AM
Subject: Re: [asterisk-users] Passing a variable from one Asterisk box
toanother
Richard Lyman wrote:
Eric Bishop wrote:
Hi all,
We currently have 2 Asterisk boxes and we pass calls to a fro. All
works
great except we now need to pass variables between them.
For example now on box 1 we have:
exten = _23XX,1,SetVar(Foo=1234)
exten = _23XX,2,Dial(SIP/${EXTEN:[EMAIL PROTECTED])
When the call dials into Box 2 the variable Foo does not get
passed...
Does anyone have any clever ideas?
as noted in asterisk/docs/README.variables (iirc)
you should see that variable inheritance can occur by prefacing the
variable with '_' or '__'
also, depending on the age of your asterisk you might want to start
using
'Set' vice 'SetVar'
also, having ${EXTEN:0} , the :0 doesn't do anything, so you should
not
use it and just have ${EXTEN}
i hope this helps
sadly replying to my own post, but, i forgot to mention that
passing variables with IAX2 can be an issue sometimes when you use
user and peer (the user side can pass vars the peer side can not, or
doesn't accept them iirc)
this does not happen using friend, but that has its own issues... check
the wiki for more thoughts about this.
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Re: [asterisk-users] BSNL caller ID (India)
On Mon, Apr 16, 2007 at 11:18:46PM +0530, Sanjay Rajdev wrote: Has anyone figured out the way of getting the caller id for BSNL on Asterisk 1.4.2 I have tried following link http://bugs.digium.com/view.php?id=6683nbn=24 but was not able to get it, although did not ge any error too. I always get the caller id as asterisk. Hmmm... are you sure you have configured your system to get callerid from the PSTN? callerid=asrecieved in zapata.conf. -- Tzafrir Cohen icq#16849755jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ${CALLERIDNUM} DEPRECATED in 1.4.2
${CALLERID(num)}
or
${CALLERID(name)}
Sanjay Rajdev wrote:
${CALLERIDNUM} is DEPRECATED in 1.4.2 what is the alternative to find the same
in extensions.conf for setting a proper dialplan.
Please Suggest
Regards,
Sanjay Rajdev
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Re: [asterisk-users] Instability on Asterisk
I don't think that this problem is DNS, becouse asterisk can send register to my provider and he can replay to asterisk, so, DNS is working fine. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, Matt [EMAIL PROTECTED]: I understand... but I know, at least in 1.2 if there was a DNS failure for some reason asterisk stopped doing anything else. That is... if I restart asterisk and it goes to register with , say, my 6 SIP upstream peers... but they are timing out for some reason asterisk won't initialize zap, or other sip or IAX stuff until it times out all 6 of those. I was under the impression this was being fixed in 1.4, but maybe it has not been. On 4/16/07, Frederico Madeira [EMAIL PROTECTED] wrote: Matt, It's make no sense. Asterisk should process messages in diferents threds, not in queue. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, Matt [EMAIL PROTECTED]: While I don't use 1.4, it could be that the registration failure (you said 100 registration lines with your provider?!?) are blocking the phones from registering. This is only a guess, I don't know for sure. On 4/16/07, Frederico Madeira [EMAIL PROTECTED] wrote: Oquendo, My provieder require sip digest authebtication: Asterisk send register to sip provider sip provider response with 401 asterisk send register again with authentication header sip provider response ok This is normal process, when problem happen, this process ocour until 401 message, and asterisk didn't add auth header. I thins this is a problem in asterisk becouse my ip phones can't register into him. After few minutes asterisk can register again and ip phones too. Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br 2007/4/16, J. Oquendo [EMAIL PROTECTED]: Frederico Madeira wrote: Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for ' [EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:18] NOTICE[2851] chan_sip.c:-- Registration for ' [EMAIL PROTECTED]' timed out, trying again (Attempt #36) [Apr 16 07:14:19] NOTICE[2851] chan_sip.c:-- Registration for ' [EMAIL PROTECTED]' timed out, trying again (Attempt #36) Sniffing netowork i saw register packets arriving from ip phones, but asterisk didn't send response to it, and for external registry i saw register sended to sip provider, 401 response from sip provider and asterisk didn't start sip digest challenger, it was send a register message again without authentication header. Network connectivity for asterisk was ok during this problem moments. My asterisk is 1.4.2 with FC6 What can be wrong ? Thanks. Why do you believe this to be an issue with Asterisk. What you describe is this in barebones YourNetworkPhones --- connect --- SIP Provider SIP Provider --- starts handshake --- Yournetwork YourNetwork --- gets ball rolling --- SIP Provider SIP Provider ... ignores you Sounds like you should be ripping into your SIP provider they're sending you unauthorized messages which sounds like either they changed something, or you did. -- J. Oquendo http://pgp.mit.edu:11371/pks/lookup?op=getsearch=0x1383A743 sil . infiltrated @ net http://www.infiltrated.net The happiness of society is the end of government. John Adams ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit:
Re: [asterisk-users] ${CALLERIDNUM} DEPRECATED in 1.4.2
Sanjay Rajdev wrote:
${CALLERIDNUM} is DEPRECATED in 1.4.2 what is the alternative to find the same
in extensions.conf for setting a proper dialplan.
Please Suggest
That information is in UPGRADE.txt -- part of the Asterisk source.
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RE: [asterisk-users] ${CALLERIDNUM} DEPRECATED in 1.4.2
${CALLERID(num)}
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sanjay
Rajdev
Sent: Monday, April 16, 2007 13:39
To: asterisk-users
Cc: asterisk-dev
Subject: [asterisk-users] ${CALLERIDNUM} DEPRECATED in 1.4.2
${CALLERIDNUM} is DEPRECATED in 1.4.2 what is the alternative to find
the same in extensions.conf for setting a proper dialplan.
Please Suggest
Regards,
Sanjay Rajdev
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[asterisk-users] Audio Problems - Operating System??
Hey All, I've been using Asterisk for a couple years now, but have always had some unsolvable audio problems. I get audio stuttering and popping quite often. Even if I have just one call up! The server is a Dual Duo-Core 3.0ghz Xeon Processor PC with 4 GB or ram. It just seems to me that this should NOT be happening. The server resources are nearly 98% idle. I've tried using the SLN audio file format, which does reduce the CPU usage when playing audio files, but it didn't help the audio quality. I've also tried putting my audio files on a RAM Drive and still have the same problem. I've also slimmed my asterisk system down to load only the modules that I am using via modules.conf. Now my question. I've heard through the grapevine that the Operating system running Asterisk can make a big difference in performance. I am currently running SuSE Linux Enterprise Server 10.A friend of mine actually talked to someone at Digium about this specific problem and they told him -not- to run SuSE. Is this correct? Has anyone else had any experience similar to this? I'm just wondering if Digium just wanted to push Asterisk Business Edition running on rPath on him, or if there really are some conflicts with SuSE that may cause audio instability. If so then it definitely would explain a lot regarding my poor audio quality problems. I would be happy to hear thoughts that any of you might have. Thanks so much! Darren Nay [EMAIL PROTECTED] ___ --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
