Russell Bauer
2005-May-19 11:14 UTC
[Asterisk-Users] Re: Asterisk-Users Digest, Vol 10, Issue 154
Send Asterisk-Users mailing list submissions to asterisk-users@lists.digium.com To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-users or, via email, send a message with subject or body 'help' to asterisk-users-request@lists.digium.com You can reach the person managing the list at asterisk-users-owner@lists.digium.com When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. Re: Grandstream ATA 286 and ilbc (Kevin McCauley) 2. Re: MusicOnHold probelms (admin) 3. RE: SIP Phone Recommendations? (Ariel Batista) 4. Re: Deleting Monitor Files After 2 Months (Steve Totaro) 5. Two TDM04 with Poweredge (Tom Hayden) 6. Re: Public vs. Private Network (Eric Wieling aka ManxPower) 7. Re: asterisk-oh323 build problems (VoIP Newbie) 8. Re: Outbound dialing issue with FXO (Johnathan Corgan) 9. RE: Two TDM04 with Poweredge (David Brodbeck) 10. RE: Re: Grandstream ATA 286 and ilbc (Anton Krall) 11. (no subject) (M O) 12. Re: Always Ringing (VoIP Newbie) 13. Re: SIP and FastStart (VoIP Newbie) 14. ACD Methods (Marshall, Ed) 15. Can't make outgoing calls (Nick Heinemans) 16. Re: Asterisk real time extensions problem... (Gentian Bajraktari) 17. AS5300 -> Meridian Configuration (Aaron Daniel) 18. Re: Phone keypad input not working during "menu's" (Don) 19. Re: Public vs. Private Network (William Suffill) 20. 3com 3101 SIP configuration (rbauer) ---------------------------------------------------------------------- Message: 1 Date: Thu, 19 May 2005 15:15:06 +0000 (UTC) From: Kevin McCauley <kevin@fullcirclenetworks.com> Subject: [Asterisk-Users] Re: Grandstream ATA 286 and ilbc To: asterisk-users@lists.digium.com Message-ID: <loom.20050519T171208-388@post.gmane.org> Content-Type: text/plain; charset=us-ascii Anton Krall <akrall-lists <at> intruder.com.mx> writes:>> >> Guys, anybody having problem with ilbc and GS ata 286? I just tried it >for >> fun (always using alaw) and voices sounded quite bad... crappy voice >> prompts, not bad quality, just like weird noises. >> >> Anybody had this? whats the latest FW for those units? >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users <at> lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > > > >Anton, I use iLBC exclusively on the 286/486 and it interoperates with other devices on my network fine. In fact I use iLBC because some of the people I talk to only have dialup and it works the best for that. I will mention though, that I have stayed on FW version 1.0.5.16 since I have had troubles with newer versions. -Kevin ------------------------------ Message: 2 Date: Thu, 19 May 2005 09:37:33 -0600 From: "admin" <dsanders@purecom.com> Subject: Re: [Asterisk-Users] MusicOnHold probelms To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <WorldClient-F200505190937.AA37330001@purecom.com> Content-Type: text/plain Do you have mpg123 installed? Is there a .mp3 file available to play in your /var/lib/asterisk/mohmp3 directory? -daryl -----Original Message----- From: chawki hammoud <cyhammoud@yahoo.com> To: Asterisk-Users@lists.digium.com Cc: Date: Thu, 19 May 2005 06:03:55 -0700 (PDT) Subject: [Asterisk-Users] MusicOnHold probelms>> This is my second attempt trying to get help and I am >> hoping someone can. When the musiconhold extension is >> matched, Asterisk attempts to execute musiconhold and >> stops right away, this is what I gets: >> >> Executing MusicOnHold("OSS/dsp", "") in new stack >> -- Started music on hold, class 'default', on >> OSS/dsp >> -- Stopped music on hold on OSS/dsp >> >> Is there a file that musiconhold try to play and can't >> find. Please help withy any suggestions. >> >> >> >> >> Discover Yahoo! >> Stay in touch with email, IM, photo sharing and more. Check it out! >> http://discover.yahoo.com/stayintouch.html >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > >------------------------------ Message: 3 Date: Thu, 19 May 2005 11:44:39 -0400 From: "Ariel Batista" <arielb27@hotmail.com> Subject: RE: [Asterisk-Users] SIP Phone Recommendations? To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <BAY104-DAV156696C5A973C8D2F9623CDB080@phx.gbl> Content-Type: text/plain; charset="us-ascii" Just want to let everyone know that even if there changing it out to the new 501 it's still on of the best. Remember that people are still buying the Cisco 7960G which is being phased out as well. The IP-500 works and works very well. I know that there price will be going down soon once there are some supplies of the IP-501. But if you need a phone now it is a very good one for the price. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Michael Graves Sent: Thursday, May 19, 2005 10:15 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SIP Phone Recommendations? On Wed, 18 May 2005 22:29:40 -0500, Kristian Kielhofner wrote:>>Ariel, >> >> It's probably not a good idea to reccomend the IP 500/300 anymore. >>They are being phased out by Polycom because they (and the IP 300) only >>have 2mb of flash, and Polycom is looking to standardize on 4mb for >>their firmware (which the IP 600 has had since day one). >> >> If you are going to buy a Polycom now, get an IP 600, or, wait forthe>>301's or 501's. Don't say I didn't warn you! > >Good advice!. BTW, I LOVE my IP600's. I also kinda like the Zultys 4x4/4x5.The hardware and software is good but their support arrangement is terrible. They provide no end user support at all. Period. They rely upon their dealers to provide all support, but then they're ok with signing up dealers that know nothing about the products. Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262 _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ------------------------------ Message: 4 Date: Thu, 19 May 2005 11:44:09 -0400 From: "Steve Totaro" <asterisk@totarotechnologies.com> Subject: Re: [Asterisk-Users] Deleting Monitor Files After 2 Months To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <001801c55c89$a50e2610$9f0aa8c0@cfigroup.computerfrontiers.com> Content-Type: text/plain; format=flowed; charset="utf-8"; reply-type=original ----- Original Message ----- From: "Matthew Boehm" <mboehm@cytelcom.com> To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Sent: Thursday, May 19, 2005 10:04 AM Subject: Re: [Asterisk-Users] Deleting Monitor Files After 2 Months>> Gavin Hamill wrote: >>> On Thursday 19 May 2005 13:51, Steve Totaro wrote: >>>> Does anyone know the best way to automate the deletion of monitor >>>> files after they age two months? >>> >>> How about ... >>> >>> $ find /path/to/files -ctime +60 -exec rm {}\; >>> >>> Cheers, >>> Gavin. >> >> Nice Gavin. I would further turn that into a shell script and pop it >into >> cron to run nightly. >> >> -Matthew >> > >Thanks! ------------------------------ Message: 5 Date: Thu, 19 May 2005 11:45:11 -0400 From: Tom Hayden <thayden@gmail.com> Subject: [Asterisk-Users] Two TDM04 with Poweredge To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <e5f28f9205051908455d4c4778@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Has anyone on this list succesfully managed to get two (or more) TDM04 (with four FXO each) working on a Dell PowerEdge server? If so, which model? Was it a hassle? I'm doing a seven-line installation and a callbank seems like overkill, I just don't want to get suck with a PowerEdge that gets into an IRQ mess. Thanks in Advance, Tom Hayden ------------------------------ Message: 6 Date: Thu, 19 May 2005 10:54:44 -0500 From: Eric Wieling aka ManxPower <eric@fnords.org> Subject: Re: [Asterisk-Users] Public vs. Private Network To: Andrew Latham <lathama@gmail.com>, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <428CB6C4.8080207@fnords.org> Content-Type: text/plain; charset=windows-1252; format=flowed>>>I am looking at connecting 7 ? 10 locations together using Asterisk and >>>possibly some VoIP gateway appliances. I need to insure best voice >quality >>>as these trunks will be used primarily for customer calls. I am >considering >>>implementing a full T1 frame relay circuit to each location which can be >>>done for a reasonable cost. DSL and Cable are currently at each >location >>>and setup for automatic failover. Should I remove one of my public >>>connections and replace it with a private circuit for best quality? > >To run VoIP over Frame Relay you need your Port Speed to be the same as your CIR. Cisco has extensive docs about this, but I'm too lazy to look them up right now. -- Always do right. This will gratify some people and astonish the rest. Mark Twain ------------------------------ Message: 7 Date: Thu, 19 May 2005 23:57:45 +0800 From: VoIP Newbie <voip.newbie@gmail.com> Subject: Re: [Asterisk-Users] asterisk-oh323 build problems To: FaberK <f.faberk@gmail.com>, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <62b5865d05051908576d725acc@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Read README file first. You will get a clue. On 5/19/05, FaberK <f.faberk@gmail.com> wrote:>> Hello Guys, >> first of all, I'm very new with asterisk. >> I'm trying to set it up. I've already compiled and installed >Asterisk-1.0.7 >> Now I'm trying with asterisk-oh323 >> I've already installed pwlib, oh323 and I've already set the variables. >> Now, when I try to "make" asterisk-oh323 I receive this error messagge: >> for x in wrapper asterisk-driver; do make -C $x all || exit 1 ; done >> make[1]: Entering directory `/root/voip/asterisk/asterisk-oh323/wrapper' >> g++ -Wall -mcpu=i586 -DP_LINUX -D_REENTRANT -DP_HAS_SEMAPHORES -DP_SSL >> -DP_PTHREADS -DPBYTE_ORDER=PLITTLE_ENDIAN -DPHAS_TEMPLATES -O3 >> -DNDEBUG -I/usr/include -I/usr/include/crypto >> -I/usr/lib/pwlib/include/ptlib/unix -I/usr/lib/pwlib/include >> -I/usr/lib/openh323/include -I../asterisk-driver -g -c wrapper.cxx -o >> wrapper.o >> wrapper.cxx: In constructor >> `WrapH323Connection::WrapH323Connection(WrapH323EndPoint&, unsigned >int, >> int, int, short unsigned int)': >> wrapper.cxx:563: `SetMaxAudioDelayJitter' undeclared (first use this >function) >> wrapper.cxx:563: (Each undeclared identifier is reported only once for >each >> function it appears in.) >> wrapper.cxx: In function `call_ret_val_t h323_clear_call(const char*)': >> wrapper.cxx:1230: warning: unused variable >`ClearCallThread*clearCallThread' >> make[1]: *** [wrapper.o] Error 1 >> make[1]: Leaving directory `/root/voip/asterisk/asterisk-oh323/wrapper' >> make: *** [subdirs_all] Error 1 >> >> >> What's wrong? >> >> Thanks >> >> -- >> .:FaberK:. >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> > > > >------------------------------ Message: 8 Date: Thu, 19 May 2005 09:12:22 -0700 From: Johnathan Corgan <jcorgan@aeinet.com> Subject: Re: [Asterisk-Users] Outbound dialing issue with FXO To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <428CBAE6.4050601@aeinet.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Mike Clark wrote:>> However, outbound calls are hit or miss. Sometimes they work fine and >> other times we get a "you must first dial a 1 or 0" message back from >> telco when dialing out standard POTS lines. > >Did you get this working yet? -Johnathan ------------------------------ Message: 9 Date: Thu, 19 May 2005 12:16:18 -0400 From: David Brodbeck <DavidB@mail.interclean.com> Subject: RE: [Asterisk-Users] Two TDM04 with Poweredge To: 'Asterisk Users Mailing List - Non-Commercial Discussion' <asterisk-users@lists.digium.com> Message-ID: <C823AC1DB499D511BB7C00B0D0F0574CC4116F@serverdell2200.interclean.com> Content-Type: text/plain; charset="iso-8859-1">> -----Original Message----- >> From: Tom Hayden [mailto:thayden@gmail.com] > > >> Has anyone on this list succesfully managed to get two (or more) TDM04 >> (with four FXO each) working on a Dell PowerEdge server? If so, which >> model? Was it a hassle? > >I've got a PowerEdge 800 tower server with two of them. Only five FXO modules right now, though. It mostly works. When I insert the driver I get an NMI, but that appears to be harmless. I have to unload and reload the drivers once a week or so, otherwise the FXO modules tend to eventually stop responding. I haven't had any audio quality or interrupt problems, though. The system gets the job done, but I can't wholeheartedly recommend these cards. If I had to do it all over again, I'd consider some other method. I'm not sure if anything else would be practical, though. A T1 card plus channel bank is kind of cost prohibitive for such a small installation. I've heard good things about the Sipura gateways, but I'm interfacing to a PBX and need the ability to flash the line for transfers, and I think Flash() is Zap-specific. ------------------------------ Message: 10 Date: Thu, 19 May 2005 11:33:29 -0500 From: "Anton Krall" <akrall-lists@intruder.com.mx> Subject: RE: [Asterisk-Users] Re: Grandstream ATA 286 and ilbc To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Message-ID: <20050519163306.839BE5B40EF@intruder.com.mx> Content-Type: text/plain; charset="us-ascii" That's what I was starting to think.. Since I've always used ulaw or alaw... Seems that firmware 1.0.5.23 has ilbc broken. |-----Original Message----- |From: asterisk-users-bounces@lists.digium.com |[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of |Kevin McCauley |Sent: Jueves, 19 de Mayo de 2005 10:15 a.m. |To: asterisk-users@lists.digium.com |Subject: [Asterisk-Users] Re: Grandstream ATA 286 and ilbc | |Anton Krall <akrall-lists <at> intruder.com.mx> writes: | |> |> Guys, anybody having problem with ilbc and GS ata 286? I |just tried it |> for fun (always using alaw) and voices sounded quite bad... crappy |> voice prompts, not bad quality, just like weird noises. |> |> Anybody had this? whats the latest FW for those units? |> |> _______________________________________________ |> Asterisk-Users mailing list |> Asterisk-Users <at> lists.digium.com |> http://lists.digium.com/mailman/listinfo/asterisk-users |> To UNSUBSCRIBE or update options visit: |> http://lists.digium.com/mailman/listinfo/asterisk-users |> |> | | |Anton, | |I use iLBC exclusively on the 286/486 and it interoperates |with other devices on my network fine. In fact I use iLBC |because some of the people I talk to only have dialup and it |works the best for that. | |I will mention though, that I have stayed on FW version |1.0.5.16 since I have had troubles with newer versions. | |-Kevin | |_______________________________________________ |Asterisk-Users mailing list |Asterisk-Users@lists.digium.com |http://lists.digium.com/mailman/listinfo/asterisk-users |To UNSUBSCRIBE or update options visit: | http://lists.digium.com/mailman/listinfo/asterisk-users | ------------------------------ Message: 11 Date: Thu, 19 May 2005 09:33:55 -0700 (PDT) From: M O <martinoshield@yahoo.com> Subject: [Asterisk-Users] (no subject) To: asterisk-users@lists.digium.com Message-ID: <20050519163355.9122.qmail@web30312.mail.mud.yahoo.com> Content-Type: text/plain; charset=us-ascii BJ,>>BJ Weschke <bweschke@gmail.com> >>Subject: Re: [Asterisk-Users] Do Both! :) Re: Telecom >>SIP termination vs. DS3 >>To: Asterisk Users Mailing List - Non-Commercial >>Discussion <asterisk-users@lists.digium.com> >>Message-ID:<79cf63305051908056c284cc9@mail.gmail.com>>>Content-Type: text/plain; charset=ISO-8859-1 > > >>Did I miss pricing/availability announcements from >>Digium on that DS3 card somewhere? > >No idea. You can contact them if you dont know what you missed :)>>I wasn't aware they were going to be GA in less than3>>weeks from now. > > >>From my standpoint, I am just so anxious andconfident that the Digium DS3 Channelized Voice PCI Card, whenever I get my order of DID #'s and test my configuration of Asterisk, that I am willing to prepay, or have available to Digium, whatever $$$ they want for the card. I am EVENTUALLY going to need it anyways, so I dont mind prepaying wheather or not it is available today! My knowledge of their product offering is no different than yours. But I fully intend on purchasing it :)! We are starting off with a 100Mbps burstable bandwith, though exspensive to start, after 30 days of usage, my bandwidth costs will look like $25K. Going off the top of head for a Sangoma DS3 Card @ $6000 per card, If I got 2 of them for $12,000 total, I eliminate, almost, that $25,000 per month bandwidth cost to me. So if Digiums DS3 Channelized Voice PCI card costs, around what Sangomas costs, $6,000, (JUST AS A EXAMPLE FOR THIS POST), $12,000 for 2 Digium DS3's in 1 month, I will save almost $10,000 AUTOMATICALLY and ever month thereafter! :) Come on Txlink DID #'s. Come on Digium with the DS3 Channelized Voice PCI card. Then all Digium would have left to do is create a board or work with someone on getting Radio Waves into your computer. :) Sincerely, SoftwareRadioGuy __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail ------------------------------ Message: 12 Date: Fri, 20 May 2005 00:36:58 +0800 From: VoIP Newbie <voip.newbie@gmail.com> Subject: [Asterisk-Users] Re: Always Ringing To: Asterisk-Users@lists.digium.com Message-ID: <62b5865d050519093621b36e6f@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Can anyone give me a big hand here?? On 5/16/05, VoIP Newbie <voip.newbie@gmail.com> wrote:>> Hi all, >> >> I am using chan_h323 from Asterisk CVS to interconnect with GNUGK >> v2.2.2. Then I made call from a H323 EP, thru GNUGK, to SIP EP on >> Asterisk. However, I only heard ringing when the call was answered on >> SIP side. Below is the debug from chan_h323. Any help is welcome. >> Thanks. >> >> *CLI> == New H.323 Connection created. >> -- Setting up Call >> -- Call token: [ip$22.7.20.32:30012/16050] >> -- Calling party name: [6907] >> -- Calling party number: [6907] >> -- Called party name: [0069777] >> -- Called party number: [0069777] >> --Received SETUP message >> =-= In OnAnswerCall for call 16050 >> - Progress Indicator: 0 >> - Inserting PI of 0 into ALERTING message >> -- Started logical channel: sending G.729 >> -- channelsOpen = 1 >> External RTP Session Starting >> RTP channel id 1 parameters: >> -- remoteIpAddress: 22.7.20.32 >> -- remotePort: 51048 >> -- ExternalIpAddress: 0.0.0.0 >> -- ExternalPort: 17816 >> -- Started logical channel: receiving G.729 >> -- channelsOpen = 2 >> External RTP Session Starting >> RTP channel id 1 parameters: >> ExternalRTPChannel Destroyed >> ExternalRTPChannel Destroyed >> -- Executing Dial("H323/ip$22.7.20.32:30012/16050", "SIP/69777") >> in new stack >> -- Called 69777 >> -- SIP/69777-c6ce is ringing >> Sending alerting >> >> -- SIP/69777-c6ce answered H323/ip$22.7.20.32:30012/16050 >> Answering call ip$22.7.20.32:30012/16050 >> -- Transmitting RFC2833 on payload 96 >> -- Received Facility message... >> =-= In OnConnectionEstablished for call 16050 >> -- Connection Established with "6907 [22.7.20.32]" >> -- Received Facility message... >> -- Started logical channel: receiving G.729 >> -- channelsOpen = 3 >> External RTP Session Starting >> RTP channel id 1 parameters: >> -- Received Facility message... >> -- Received RELEASE COMPLETE message... >> -- ClearCall: Request to clear call with token >> ip$22.7.20.32:30012/16050, cause EndedByRemoteUser >> -- Sending RELEASE COMPLETE >> channelsOpen = 2 >> channelsOpen = 1 >> channelsOpen = 0 >> ExternalRTPChannel Destroyed >> ExternalRTPChannel Destroyed >> ExternalRTPChannel Destroyed >> -- ClearCall: Request to clear call with token >> ip$22.7.20.32:30012/16050, cause EndedByTransportFail >> == Spawn extension (default, 0069777, 1) exited non-zero on >> 'H323/ip$22.7.20.32:30012/16050' >> -- 6907 [22.7.20.32] has cleared the call >> == H.323 Connection deleted. >> > > > >------------------------------ Message: 13 Date: Fri, 20 May 2005 00:38:46 +0800 From: VoIP Newbie <voip.newbie@gmail.com> Subject: [Asterisk-Users] Re: SIP and FastStart To: Asterisk-Users@lists.digium.com Message-ID: <62b5865d0505190938e93b184@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1 Can anyone give me a big here? On 5/13/05, VoIP Newbie <voip.newbie@gmail.com> wrote:>> I am using Asterisk-oh323 v0.7.1 with GNUGK. Please advise what must >> be done to make FastStart work with SIP phones. Thanks. >> >> On 5/12/05, VoIP Newbie <voip.newbie@gmail.com> wrote: >> > Hi all, >> > >> > When I enabled faststart in oh323.conf, calls from H323 endpoint to >> > SIP phones could not complete. The originating phone kept ringing when >> > calls were answered by SIP phones. >> > >> > fastStart=yes >> > h245Tunnelling =yes >> > h245inSetup=yes >> > >> > Please can you advise. >> > >> > Many Thanks. >> > >> > > > >------------------------------ Message: 14 Date: Thu, 19 May 2005 17:43:52 +0100 From: "Marshall, Ed" <ed.marshall@cetelem.co.uk> Subject: [Asterisk-Users] ACD Methods To: "'asterisk-users@lists.digium.com'" <asterisk-users@lists.digium.com> Message-ID: <8ADF3F2FBF4F2C408C4C47B05CA3D267121A0C@HCCEXC02> Content-Type: text/plain; charset="iso-8859-1" Can anyone point me in the right direction of info regarding ACD methods available in Asterisk. As far as I can see there are time based ring strategies available but I cannot find any info regarding skills based routing or queue priorities. Also do the current time based ring strategies work globally. What I mean by this is if an agent is a member of more than one queue then would the ACD algorithm take this into account before deciding to allocate another call ? Any help would be much appreciated. Regards Ed ------------------------------ Message: 15 Date: Thu, 19 May 2005 18:46:42 +0200 From: "Nick Heinemans" <nick@heinemans.net> Subject: [Asterisk-Users] Can't make outgoing calls To: <asterisk-users@lists.digium.com> Message-ID: <20050519164640.848BC18DE8@olive.qinip.net> Content-Type: text/plain; charset="us-ascii" Hello, When I try to make an outgoing call from my X-lite softphone connected to Asterisk, I keep getting the following error message: May 19 18:42:58 WARNING[3086]: Forbidden - wrong password on authentication for INVITE to '"31307110340" <sip:31307110340@84.41.149.228>;tag=as13ba1ff7' I'm running AAH 1.0 on a server which is directly hooked up to my ADSL line. It's second NIC is connected to my LAN on which the PC with X-lite is also connected. I've configured the Asterisk server as a NAT router and I opened UDP ports 5060 and 10000-20000 from the outside. Any idea what might be wrong? Regards, Nick ------------------------------ Message: 16 Date: Thu, 19 May 2005 17:50:38 +0100 From: "Gentian Bajraktari" <g.bajraktari@afb.net.al> Subject: Re: [Asterisk-Users] Asterisk real time extensions problem... To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <00c301c55c92$e7051de0$0300a8c0@mozart> Content-Type: text/plain; charset="iso-8859-1" HI, The problem is that you are using: incoming-next,60069,1 Use: incoming-next|60069|1 instead RG, Gentian ----- Original Message ----- From: Bharat M. Sarvan To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Cc: asterisk-dev@lists.digium.com Sent: Sunday, April 17, 2005 11:52 AM Subject: [Asterisk-Users] Asterisk real time extensions problem... Hello everybody, I have setup asterisk real time extensions and its working pretty well. But the problem is when I am jumping between the contexts using the Goto statement in the database. I am getting a error = Parsing '/etc/asterisk/sip_notify.conf': Found -- SIP Seeding peers from Astdb: 'ezzibpo4' at ezzibpo4@210.211.246.47:5061 for 60 -- Executing Goto("SIP/ezzibpo4-4636", "incoming-next,6069,1") May 19 05:00:04 NOTICE[6420]: pbx.c:1688 pbx_extension_helper: Cannot find extension '6069' in context 'incom' May 19 05:00:04 WARNING[6420]: pbx.c:6256 ast_parseable_goto: Priority 'incoming-next, The structure of the extensions db is as given below +----+---------------+-------+----------+-----------------+----------------------+ | id | context | exten | priority | app | appdata | +----+---------------+-------+----------+-----------------+----------------------+ | 1 | incoming | 6069 | 1 | Goto | incoming-next,6069,1 | | 2 | incoming | 6069 | 2 | Hangup | | | 3 | incoming-next | 6069 | 1 | DigitTimeout | 10 | | 4 | incoming-next | 6069 | 2 | ResponseTimeout | 30 | | 5 | incoming-next | 6069 | 3 | Background | welcome | The context "incom" in the above error is the context defined for placing outgoing call in the sip.conf file. I don't understand why is it looking for extension 6069 in the "incom" context. The "Goto" statement in the context incoming is getting executed without any probs, but the control is not getting transferred to the context "incoming-next" upon execution of the Goto statement. Could anybody suggest me as to where might the problem be and any way to get rid of this problem. Please do reply.. Regards, Bharat M. Sarvan ------------------------------------------------------------------------------ _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050519/68eec2d3/attachment.html ------------------------------ Message: 17 Date: Thu, 19 May 2005 11:51:15 -0500 From: Aaron Daniel <amdtech@shsu.edu> Subject: [Asterisk-Users] AS5300 -> Meridian Configuration To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <ED6ACD5A-E677-43A4-A1E4-1502ABACF545@shsu.edu> Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed We're trying to set up a connection between an AS5300 and a meridian CSU/DSU so our asterisk system can interconnect with our current legacy system, and for some reason the T1 connection will not come up whatsoever. I've gone through all the configurations I can think of, even basically copied our current cisco settings directly to the AS5300 so they would be nearly identical, and nothing. Any help would be appreciated. AS5300 config: Current configuration : 2094 bytes ! ! Last configuration change at 11:45:56 CDT Thu May 19 2005 ! NVRAM config last updated at 11:46:14 CDT Thu May 19 2005 ! version 12.3 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname cdial2 ! boot-start-marker boot-end-marker ! enable secret 5 ******************* enable password ******************* ! spe 1/0 1/7 firmware location system:/ucode/mica_port_firmware spe 2/0 2/7 firmware location system:/ucode/mica_port_firmware ! ! resource-pool disable clock timezone CST -6 clock summer-time CDT recurring ! no aaa new-model ip subnet-zero no ip routing ip finger ip domain name shsu.edu ip name-server 158.135.1.20 ip name-server 158.135.1.200 ! ! isdn switch-type primary-5ess isdn voice-call-failure 0 ! ! ! ! ! ! ! ! ! ! ! ! controller T1 0 shutdown framing sf linecode ami ! controller T1 1 shutdown framing sf linecode ami ! controller T1 2 framing esf clock source line primary linecode b8zs pri-group timeslots 1-24 ! controller T1 3 shutdown framing sf linecode ami ! ! interface Ethernet0 no ip address no ip route-cache shutdown ! interface Serial2:23 no ip address ip mroute-cache dialer-group 1 isdn switch-type primary-5ess isdn protocol-emulate network isdn incoming-voice modem isdn disconnect-cause 1 fair-queue 64 16 3 no cdp enable ip rsvp bandwidth ip rtp reserve 10000 10000 ! interface FastEthernet0 ip address 158.135.1.61 255.255.0.0 no ip route-cache no ip mroute-cache duplex full speed 100 no mop enabled ! ip classless no ip http server ! ! ! ! ! ! ! dial-peer voice 6 voip incoming called-number 6.... destination-pattern 6.... session protocol sipv2 session target sip-server ! dial-peer voice 4 pots application session direct-inward-dial ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server dns:sipproxy1.shsu.edu ! ! line con 0 line 1 96 line aux 0 line vty 0 4 password ******************* login ! scheduler interval 1000 ntp clock-period 17180204 ntp update-calendar ntp server 158.135.1.2 ! end meridian config: ADAN DCH 13 CTYP MSDL GRP 0 DNUM 6 PORT 3 DES ASVOIP USR PRI DCHL 25 OTBF 32 PARM RS422 DTE DRAT 64KC CLOK EXT IFC ESS5 SIDE USR CNEG 1 RLS ID 36 RCAP ND2 T200 3 T203 10 N200 3 N201 260 K 7 Thanks, Aaron Daniel Senior Voice Analyst Sam Houston State University ------------------------------ Message: 18 Date: Thu, 19 May 2005 09:54:36 -0700 From: Don <asterisk@geeksrus.ca> Subject: Re: [Asterisk-Users] Phone keypad input not working during "menu's" To: asterisk-users@lists.digium.com Message-ID: <428CC4CC.8080502@geeksrus.ca> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Wilson Pickett wrote:>>What codec are your phones using and which do you have in sip.conf in >>general and phone entries? >> >> > >Hi Wilson, Thanks for the reply. I didn't know anything about codecs but I've tried to look up what I can. The Polycom documentation (SIP admin guide) says the hone supports G.711u-law, G.711a-law, G.729AB, SID and RFC2833. The phone configuration files say that the preference is for u-law, a-law and AB in that order. My sip.conf file says: disallow=all allow=ulaw allow=alaw I would guess that means I'm ok (i.e. ulaw is good on both sides) but this is a new area of * for me. What do you think? Don ------------------------------ Message: 19 Date: Thu, 19 May 2005 12:56:22 -0400 From: William Suffill <william.suffill@gmail.com> Subject: Re: [Asterisk-Users] Public vs. Private Network To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <6b65470d05051909562a778d83@mail.gmail.com> Content-Type: text/plain; charset=WINDOWS-1252 Point to Point connectivity if they are close enough. Only use DSL/Cable if you have to since results may vary depnding on location/route/utilization/ISP. On 5/19/05, Andrew Latham <lathama@gmail.com> wrote:>> yes >> >> On 5/19/05, David Sampson <dsampson@innseason.com> wrote: >> > >> > >> > >> > Hello ? >> > >> > >> > >> > I am looking at connecting 7 ? 10 locations together using Asterisk >and >> > possibly some VoIP gateway appliances. I need to insure best voice >quality >> > as these trunks will be used primarily for customer calls. I am >considering >> > implementing a full T1 frame relay circuit to each location which can >be >> > done for a reasonable cost. DSL and Cable are currently at each >location >> > and setup for automatic failover. Should I remove one of my public >> > connections and replace it with a private circuit for best quality? >> > >> > Thank you, >> > >> > >> > Dave >> > >> > >> > _______________________________________________ >> > Asterisk-Users mailing list >> > Asterisk-Users@lists.digium.com >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> > To UNSUBSCRIBE or update options visit: >> > >> > http://lists.digium.com/mailman/listinfo/asterisk-users >> > >> > >> >> >> -- >> <sig> >> Andrew Latham - AKA: LATHAMA (lay-th-ham-eh) >> WWW: http://lathama.com >> Email: lathama@lathama.com - lathama@yahoo.com - lathama@gmail.com >> If any of the above are down we have bigger problems than my email! >> </sig> >> >> _______________________________________________ >> Asterisk-Users mailing list >> Asterisk-Users@lists.digium.com >> http://lists.digium.com/mailman/listinfo/asterisk-users >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >> >> > >------------------------------ Message 20: Is there anyone that has been able to get a 3com IP phone to function with asterisk. I have a couple of 3com 3101's and I'm stuck trying to figure out what I'm missing. There really shouldn't be a big difference between the Cisco 7960's and my 3com 3101, should there be? As long as it's SIP compliant? This is my first time posting out here so if this posting has "newbie" written all over it, it's true. The important thing is that I'm very interested and excited about Asterisk as are all of you. Thanks in advance. _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users End of Asterisk-Users Digest, Vol 10, Issue 154 ***********************************************