Hi What codec are you using? Best to use iLBC, 711U/A caused the same problem with our system. What handsets are you using? Grandstream work well with iLBC firmware ver.11. The problem is that there are not to many phones that work well with iLBC.
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Venu V Sent: Sunday, November 14, 2004 3:13 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Music On Hold Problem Recently I have configured Music On Hold option in asterisk PBX. But I am unable to listen to the audio properly and morever its getting breaks for every 3 seconds. If any one know about this. Please help me Thanks & Regards V.Venu -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Sent: Sunday, November 14, 2004 12:07 PM To: [EMAIL PROTECTED] Subject: Asterisk-Users Digest, Vol 4, Issue 181 Send Asterisk-Users mailing list submissions to [EMAIL PROTECTED] 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 [EMAIL PROTECTED] You can reach the person managing the list at [EMAIL PROTECTED] When replying, please edit your Subject line so it is more specific than "Re: Contents of Asterisk-Users digest..." Today's Topics: 1. RE: SysMaster and GPL Violation (Brian) 2. Re: getting callerid from spa3k to asterisk (Randy Bush) 3. my asterisk drops connection when remote side puts me on hold? (Steve Prior) 4. Cisco ATA and G729 (kido noagbodji) 5. Remote answer not detected (DB) 6. Re: SysMaster and GPL Violation (Voip Business) 7. RE: Cable for T1 connection: Crossover or straightthrough? (Franceen Thompson) 8. RE: Cisco ATA and G729 (Franceen Thompson) 9. Queue/AgentCallbackLogin Problems (Franceen Thompson) ---------------------------------------------------------------------- Message: 1 Date: Sat, 13 Nov 2004 19:30:06 -0700 From: "Brian" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] SysMaster and GPL Violation To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="us-ascii" > > Are you saying that those of us that are using the product should not be > > allowed to voice our opinions about its licensing, development and > > maintenance? That we should all just shut up and take whatever Mark & > > co. give us? If that's the case, then this is most definitely NOT an > > open-source project at all. > > > -----Original Message----- > From Brandon Patterson > Sent: Saturday, November 13, 2004 7:15 PM > Uh ok...So when will Asterisk be a licensed product? Will it take the > form of a Redhat sort of platform... Fedora & with Redhat the pay me money > side of the house? > > Just a simple question: When can we expect to see Asterisk the licensed > as in paid for version ? > > > Brandon Right now. As far as I know, you just need to contact Digium's sales department and negotiate a licensing agreement with them. ------------------------------ Message: 2 Date: Sat, 13 Nov 2004 19:11:10 -0800 From: Randy Bush <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Re: getting callerid from spa3k to asterisk To: splatters <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii > if i have two sip contexts for my spa3k, on inbound and > one outbound, e.g. > > [spa3k-in] > type=friend > host=dynamic > port=5061 > auth=md5 > secret=pfui > qualify=1000 > canreinvite=yes > context=ext-in42 > > [spa3k-out] > type=peer > auth=md5 > secret=pfui > username=outpass > fromuser=outpass > host=spa3k.bogus.com > port=5061 > nat=no > canreinvite=yes > context=ext-in42 > > and the spa3k's PSTN / Subscriber Information / User ID: = spack-in, > > the incoming connection from spa3k to * is being routed to the > spa3k-out context, not the spa3-in context. see appended. > > i suspect this is a bug in * 1.0.1. i found the problem, or at least a work-around. if i reverse the order of the above two sip contexts, the incoming call is properly routed to the spa3k-in sip context as opposed to the wrong one, spa3k-out. my guess is that * is traversing a list and taking the first context which has the ip address and port it wants without checking the context name against the name which was received over the wire. so it depends on what order the contexts are inserted in the list. aiiiiiiiiiiiiiiiiiiiiii! randy ------------------------------ Message: 3 Date: Sat, 13 Nov 2004 22:33:58 -0500 From: Steve Prior <[EMAIL PROTECTED]> Subject: [Asterisk-Users] my asterisk drops connection when remote side puts me on hold? To: Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii; format=flowed I've got a TDM100P card with a fxo and fxs module in the US. I'm using kewlstart for all ports. I've noticed that when I make a call out from an analog phone out the POTS line that if after talking to the party I called (in this case the phone company itself) they put me on hold asterisk disconnects the call immediatly. I've looked around the web pages, but can't figure out what might be causing this and how to fix it - can anyone give me a clue? Thanks Steve Prior ------------------------------ Message: 4 Date: Sun, 14 Nov 2004 03:36:55 -0000 From: "kido noagbodji" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Cisco ATA and G729 To: <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="iso-8859-1" Hi all, I am new to asterisk. I was able, but not without pain to install it on a FreeBSD box. I set up a cisco ATA 186 and the SJlabs softphone to work with the PBX. Three remarks: * On the SJphone, i use the GSM and the G711 (ulaw and alaw) codec. In the h323.conf file i enabled those codec. Everything works great!!! * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or three keys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? * Also when i set my ATA codec to g729 and in asterisk i allow=g729, i get a very low weird sound. What is that due to? I am guessing that i don't have the codec installed on the system. Is there an open source g729 codec available for FreeBSD? Any help will be very much appreciated, Thanks. Kido -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041114/f7 5233a0/attachment-0001.html ------------------------------ Message: 5 Date: Sat, 13 Nov 2004 22:45:13 -0500 From: DB <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Remote answer not detected To: [EMAIL PROTECTED] Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=us-ascii; format=flowed > > Il dom, 2004-11-14 alle 00:13, DB ha scritto: >> Here's my a section of my simple extensions.conf > <snip> >> exten => s,5,Dial(Zap/4/2326932|15) >> exten => s,6,Voicemail,u100 > <snip> >> It works, but when the call is routed out on ZAP/4 (at priority 5), >> Asterisk seems to not realize the call is answered. After 15 seconds it >> proceeds to voicemail interrupting the call. Can anyone help? > > eh, perhaps with some details about your zap... > ie what card? > zaptel.conf? > zapata.conf? > > matteo, still without divinatory powers Hi - thanks for the reply - here's that info: card is TDM22B zaptel.conf: ================== fxoks=1-2 # Make sure that the FXS(green) modules are closest fxsks=3-4 # This is for the FXO module(s) becaus defaultzone=us loadzone=us ================== zapata.conf: =================== [trunkgroups] [channels] switchtype=national signalling=fxo_ls rxwink=300 ; Atlas seems to use long (250ms) winks usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=0.0 txgain=0.0 group=1 callgroup=1 pickupgroup=1 immediate=no busydetect=yes callprogress=yes progzone=us signalling=fxo_ks echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived group=1 context=MD_line1 ; Points to the default context of your extensions.conf channel => 1 signalling=fxo_ks echocancel=yes ; You can set this to 32, 64, or 128, tweak to your needs. echocancelwhenbridged=yes echotraining=400 ; Asterisk trains to the beginning of the call, number is in milliseconds callerid=asreceived group=2 context=MD_line2 ; Points to the default context of your extensions.conf channel => 2 signalling=fxs_ks group=3 context=incoming_9141252 channel=> 3 ; Again if you only have one FXO module remove the '-4' signalling=fxs_ks group=4 context=incoming_3493729 channel=> 4 ; Again if you only have one FXO module remove the '-4' ================================ DB ------------------------------ Message: 6 Date: Sat, 13 Nov 2004 23:03:49 -0600 From: Voip Business <[EMAIL PROTECTED]> Subject: Re: [Asterisk-Users] SysMaster and GPL Violation To: Asterisk Users Mailing List - Non-Commercial Discussion <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset=US-ASCII Guys, in fact we will give an Applause to Sysmaster guys that are doing a great job in their products (world wide sales), this guys are doing money , and in this point is ,, Mark (and/or Digium is not receiving money for that). But Back to basics of Open source you can sale , modify , distribute and so on , so on so on,, BUT give the credits of the developer (in this case that is the only thing thery are not respecting) Now ,, Guys this is an oportunity to have a back benefit of that ,, because why I will pay a 150K usd for a NORFA (sysmaster new system)if for much less I can have an Asterisk up and running, In my point of view technically Asterisk (as it is right now) is GREAT lets take advance of that ,, instead of 100's of brains trying to make asterisk more and more and more features.... GUYS FOR GOD SAINT lets do it NEAT (GUI, Administration etc) Why dont Asterisk comunity opens a group for Asterisk Simplification This is the Opinion of a Non guru fellow ben available for a monthly donation for that proyect (what we need is to be Several like me to pay good programers and develop a GUI) and off course with a "licencing" that for all the donators and contributors the GUI has no Cost (including Digium) but for others will have a cost. AGAIN , this is only my point of view and I respect every coments about this :) Regards Humberto On Sat, 13 Nov 2004 19:30:06 -0700, Brian <[EMAIL PROTECTED]> wrote: > > > > > Are you saying that those of us that are using the product should not be > > > allowed to voice our opinions about its licensing, development and > > > maintenance? That we should all just shut up and take whatever Mark & > > > co. give us? If that's the case, then this is most definitely NOT an > > > open-source project at all. > > > > > -----Original Message----- > > From Brandon Patterson > > Sent: Saturday, November 13, 2004 7:15 PM > > Uh ok...So when will Asterisk be a licensed product? Will it take the > > form of a Redhat sort of platform... Fedora & with Redhat the pay me money > > side of the house? > > > > Just a simple question: When can we expect to see Asterisk the licensed > > as in paid for version ? > > > > > > Brandon > > Right now. > > As far as I know, you just need to contact Digium's sales department and > negotiate a licensing agreement with them. > > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ------------------------------ Message: 7 Date: Sat, 13 Nov 2004 23:13:37 -0700 From: "Franceen Thompson" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Cable for T1 connection: Crossover or straightthrough? To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="Windows-1252" You've got a 50/50 shot. Try the crossover. http://www.cisco.com/en/US/products/hw/routers/ps233/products_tech_note0 9186a00800a3f09.shtml#topic2 It would be more helpful for you to send your /etc/zaptel.conf file and /etc/asterisk/Zapata.conf file. You should have something like the following for your zaptel.conf file: #zaptel.conf span=1,1,0,esf,b8zs loadzone = us defaultzone=us also do an %asterisk -r and send the info from the CLI that is show when you try to dial something. It's pretty intuitive output. Tim. > -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of Andrew Kohlsmith > Sent: Saturday, November 13, 2004 6:09 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] Cable for T1 connection: Crossover or > straightthrough? > > On November 13, 2004 12:11 pm, Malcolm Bader wrote: > > This is my second asterisk server but the first one with a T100P card. > > I connected it to the phone company(SBC) jack but have only a busy > > signal when calling the T1's number and nothing in the asterisk log > > files to indicate a connection. > > Do I need to use a crossover cable? > > Is the T100P's light Green or flashing red? > > Green means it sees the other side, and the other side sees it. i.e. the > cabling is fine. Orange (well it's trying to be yellow) means that the > other > side can't see the T100P, but the T100P is seeing the other side. > > -A. > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > --- > Incoming mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com). > Version: 6.0.794 / Virus Database: 538 - Release Date: 11/10/2004 > --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.794 / Virus Database: 538 - Release Date: 11/10/2004 ------------------------------ Message: 8 Date: Sat, 13 Nov 2004 23:17:37 -0700 From: "Franceen Thompson" <[EMAIL PROTECTED]> Subject: RE: [Asterisk-Users] Cisco ATA and G729 To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="windows-1252" I'm not sure about the G711 codec on the ATA, but I know you need to purchase the g729 from digium. HYPERLINK "http://www.digium.com/index.php?menu=asterisk_g729"http://www.digium.co m/index.php?menu=asterisk_g729 pretty inexpensive at $10 each. That's for "concurrent" connections to the server. Tim. -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of kido noagbodji Sent: Saturday, November 13, 2004 8:37 PM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Cisco ATA and G729 Hi all, I am new to asterisk. I was able, but not without pain to install it on a FreeBSD box. I set up a cisco ATA 186 and the SJlabs softphone to work with the PBX. Three remarks: * On the SJphone, i use the GSM and the G711 (ulaw and alaw) codec. In the h323.conf file i enabled those codec. Everything works great!!! * However, when i set my Cisco ATA to G711, i can't hear any sound unless I press at least two or three keys(any random keys). I am using the demo context of extension.conf file. Can that be due to a fast start problem? Anyone knows how to checkthe faststartcapabilities of an ATA 186? * Also when i set my ATA codec to g729 and in asterisk i allow=g729, i get a very low weird sound. What is that due to? I am guessing that i don't have the codec installed on the system. Is there an open source g729 codec available for FreeBSD? Any help will be very much appreciated, Thanks. Kido --- Incoming mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.794 / Virus Database: 538 - Release Date: 11/10/2004 --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.794 / Virus Database: 538 - Release Date: 11/10/2004 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041113/51 deb48c/attachment-0001.html ------------------------------ Message: 9 Date: Sat, 13 Nov 2004 23:36:59 -0700 From: "Franceen Thompson" <[EMAIL PROTECTED]> Subject: [Asterisk-Users] Queue/AgentCallbackLogin Problems To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <[EMAIL PROTECTED]> Message-ID: <[EMAIL PROTECTED]> Content-Type: text/plain; charset="windows-1252" I am having a few problems with my queue. I am using the AgentCallbackLogin feature. When the call comes to the user, it does not "announce" the call to the agent. It waits until you enter the "#". After you hit #. It will play the queue-support announcement to the agent and tell them to press # if they want the call. The agent will have to hit # multiple times if they want the call to come through. Any suggestions? Tim. Extenstions.conf exten => 999,1,AgentCallbackLogin(${CALLERIDNUM}|@default) exten => 999,2,Hangup exten => 1,1,Playback(welcome) exten => 1,2,SetVar(QUEUE_PRIO=10) exten => 1,3,Queue(cssupport|t||queue-support|120) Queues.Conf [cssupport] music = random announce = queue-support strategy = rrmemory context = exitqueue timeout = 45 retry = 10 wrapuptime=60 maxlen = 2 announce-frequency = 60 announce-holdtime = yes announce-round-seconds = 10 queue-youarenext = queue-youarenext queue-thereare = queue-thereare queue-callswaiting = queue-callswaiting queue-holdtime = queue-holdtime queue-minutes = queue-minutes queue-seconds = queue-seconds queue-thankyou = queue-thankyou queue-lessthan = queue-less-than joinempty = no leavewhenempty = yes member => Agent/309 member => Agent/311 CLI> -- Executing Playback("Zap/1-1", "welcome") in new stack -- Playing 'welcome' (language 'en') -- Executing SetVar("Zap/1-1", "QUEUE_PRIO=10") in new stack -- Executing Queue("Zap/1-1", "cssupport|t||queue-support|120") in new stack -- Started music on hold, class 'random', on Zap/1-1 -- Stopped music on hold on Zap/1-1 -- Playing 'queue-youarenext' (language 'en') -- Told Zap/1-1 in cssupport their queue position (which was 1) -- Playing 'queue-thankyou' (language 'en') -- Started music on hold, class 'random', on Zap/1-1 Nov 13 23:28:26 NOTICE[4125]: app_queue.c:749 wait_for_answer: No one is answering queue 'cssupport' Nov 13 23:28:36 NOTICE[4125]: app_queue.c:749 wait_for_answer: No one is answering queue 'cssupport' Nov 13 23:28:47 NOTICE[4125]: app_queue.c:749 wait_for_answer: No one is answering queue 'cssupport' Nov 13 23:28:57 NOTICE[4125]: app_queue.c:749 wait_for_answer: No one is answering queue 'cssupport' Nov 13 23:29:08 NOTICE[4125]: app_queue.c:749 wait_for_answer: No one is answering queue 'cssupport' --- Outgoing mail is certified Virus Free. 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