[asterisk-users] Re: Sangoma A200D and DTMF Detection
We experienced this problem with a Sangoma A104D card. With echo cancel turned on, the card was not detecting incoming DTMF digits to our IVR properly. However, when we added the line relaxdtmf=yes to zapata.conf, the problem went away. If the relaxdtmf setting is not curing the problem for you, I would suggest sending an email to [EMAIL PROTECTED]. Rana Dutt Softel Solutions rdutt at softelinc dot 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] Polycom phone cycles between UNREACHABLE and REACHABLE
I have a customer witha Polycom 501 phone behind a NAT. His phone is connected tohis Netgear router at home which in turn is connected to his cable modem. The phone is set up to register with our remote Asterisk server which is on a public, static IP address, with no NAT. If we set qualify=yes, our Asterisk console shows his extension becoming UNREACHABLE for a minute, then REACHABLE for a minute, then UNREACHABLE again, in an endless cycle. If we try to call the phone while it is UNREACHABLE, the phone never rings and the call goes straight to voice mail.This is very annoying. If we set qualify=no, then if we try to call the phone, the phone sometimes does not ring at all, and we hear silence. The call eventually goes to voice mail. This is equally annoying to the customer. What is the solution to this problem? We have other customers with Polycom phones behind NAT, and they don't have this problem. Will we have better luck if we replace the Polycom with a Linksys 942 phone? Here is some console output: Jul 16 21:44:24 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE! Last qualify: 174Jul 16 21:45:33 NOTICE[19981]: chan_sip.c:9697 handle_response_peerpoke: Peer '280' is now REACHABLE! (3181ms / 5000ms) Jul 16 21:47:37 NOTICE[19981]: chan_sip.c:11364 sip_poke_noanswer: Peer '280' is now UNREACHABLE! Last qualify: 175 Here is the way the phone is set up in sip.conf: [280]type=peerusername=280secret=280host=dynamicdtmfmode=rfc2833callerid=John 280context=company_xmailbox=280nat=yescanreinvite=noqualify=5000We are using Asterisk 1.2.5 with standard .conf files. We are not using realtime or databases. Any help would be highly appreciated. Rana Dutt Softel Solutions [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] Jittery Linksys/Sipura meetme conference fixed
Ina previous message I described how Linksys 942 phone users who dialed in to a meetme conference at their site heard severe jitter. This was also experienced with Sipura SPA-2002 ATAs. Users of other IP phones like Polycom and Snom had no such problem. Also, the Linksys and SPA users had no problems with regular phone calls, just the meetme conference. This problem was finally fixed by going in to the settings for the Linksys phone and setting the RTP frame length to 0.020 and disabling the jitter buffer adjustment. The same fix also worked for the SPA. Hope this helps others who have experienced a similar problem. Rana Dutt Softel Solutions www.softelinc.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] Jittery meetme conference using Linksys 942 phones
We have two Linksys 942 phones which sound great when they call each other directly through Asterisk. But when they both dial in to a meetme conference room, the sound is very jittery. Other phones like Polycom 501 and Snom 360 sound fine when using meetme. Both Linksys phones are set to use the default g711u (ulaw) codecs. Adjusting the jitter buffer and jitter level settings to various values did not help. We are running Asterisk 1.2.1 on Centos 4.2 (Linux 2.6x kernel) on a dual-processor Dell Poweredge 2850 server with 1 Gb RAM. This machine has a TE-210 Dual-T1 card plugged in. The meetme.conf file has no general settings, just a list of two conference rooms. Has anyone else experienced sound quality issues with meetme conferences using Linksys phones? Any idea what could fix this? Thanks. Ron ___ --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 phone failover using DNS SRV?
Has anyone successfully had a SIP phone fail over from Asterisk Server A to Server B using DNS SRV? If so, which phone worked for you? I'm assuming you set up your DNS SRV records so that the IP addresses of A and B are associated with the same name, and both servers have equal priority and equal weight. In order to make calls through B after A goes down, do you have to wait as long as the registration retry interval? Or can you make calls through B as soon as you pick up the phone and dial, because the INVITE message through A fails, and the phone re-sends the INVITE through B? Thanks for any help. We've been trying this with Aastra 480i phones and SJ Phone without much luck so far. Rana ___ 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
Re: [Asterisk-Users] Enabling rtcachefriends prevents phones from calling each other
Kevin P. Fleming wrote: Matthew Boehm wrote: Can't it be changed so that if Server A has stored an unknown address for phone B that if it needs to contact B again it should look up in the database to try and contact it instead of just giving up? Perhaps rtagressive option? Contact only, not storing info in cache. Can it be done? Of course, it's all just code :-) I think it would be reasonable to add that as an option, but the number of Realtime-related options is rapidly getting out of hand and people will not be able to understand what they all do and how they interact... Please, please add this option. If you send me a patch, I will gladly volunteer to test it thoroughly. Having both MWI working and multiple servers working is a must for us. Thanks much, Rana Dutt ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Quintum Tenor DX
Has anyone used a Quintum Tenor DX to send SIP calls from Asterisk to T1 trunks and vice versa? If this works, there may be some advantages over plugging a Digium Quad T1 card directly into the * server. For example, it offloads the * server from doing the SIP to TDM conversion, which may result in a higher number of calls per second. Also, if the * server dies, a hot spare could take over and connect to the Tenor to keep the T1 connectivity up. The only cons I can think of are the increased cost of the Tenor compared to the Quad T1 card and the increased traffic on the LAN. But at a site that requires 8 T1s, the improved throughput and redundancy could outweigh these cons. Rana Dutt ___ 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
[Asterisk-Users] Problem with AstTapi
I wanted to use Outlook 2000 to dial my Contacts using Asterisk. So I installed AstTapi on my Windows XP machine. When I try to dial a contact, the call originates just fine. My SIP phone rings, and when I pick up, Asterisk makes the call to the dialed number correctly. However, Outlook displays an error message saying Unable to complete an operation requested by the automatic phone dialer. Please make sure your modem, phone and phone line are properly configured. After closing the error message dialog, if I then go to dial the Contact again, I get a different error message saying An internal error occurred in the phone dialer. Close the Dial Phone dialog box and then open it again. Well, closing the dialog box and opening it again doesn't work: the same internal error message keeps popping up when trying to make a call. The only way to get rid of it is to exit Outlook and restart it. Has anyone who has used AstTapi seen this problem? I am using Outlook 2000 SP3. My Asterisk TAPI driver is configured as follows: Host: 192.168.2.11 (IP of Asterisk server) Port: 5038 Dial out by using the Dial application - Outgoing chan: Zap/1/ User: john Password: my_secret User channel: SIP/200 My manager.conf is as follows: [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [john] secret = mysecret deny=0.0.0.0/0.0.0.0 permit=192.168.2.17/255.255.255.0 read = system,call,log.verbose,command,agent,user write = system,call,log.verbose,command,agent,user As I said, the first time I place the call from Outlook, it works fine. The trace on Asterisk shows: == Manager 'john' logged on from 192.168.2.17 -- Launching Dial(Zap/1/18005551212) on SIP/200-da5d -- Called 1/18005551212 == Manager 'john' logged off from 192.168.2.17 -- Zap/1-1 answered SIP/200-da5d -- Hungup 'Zap/1-1' Any help would be much appreciated. Rana Dutt Softel, Inc. ___ 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
[Asterisk-Users] Outlook reports internal error after using AstTapi
I wanted to use Outlook 2000 to dial my Contacts using Asterisk. So I installed AstTapi on my Windows XP machine. When I try to dial a contact, the call originates just fine. My SIP phone rings, and when I pick up, Asterisk makes the call to the dialed number correctly. However, Outlook displays an error message saying Unable to complete an operation requested by the automatic phone dialer. Please make sure your modem, phone and phone line are properly configured. After closing the error message dialog, if I then go to dial the Contact again, I get a different error message saying An internal error occurred in the phone dialer. Close the Dial Phone dialog box and then open it again. Well, closing the dialog box and opening it again doesn't work: the same internal error message keeps popping up when trying to make a call. The only way to get rid of it is to exit Outlook and restart it. Has anyone who has used AstTapi seen this problem? I am using Outlook 2000 SP3. My Asterisk TAPI driver is configured as follows: Host: 192.168.2.11 (IP of Asterisk server) Port: 5038 Dial out by using the Dial application - Outgoing chan: Zap/1/ User: john Password: my_secret User channel: SIP/200 My manager.conf is as follows: [general] enabled = yes port = 5038 bindaddr = 0.0.0.0 [john] secret = mysecret deny=0.0.0.0/0.0.0.0 permit=192.168.2.17/255.255.255.0 read = system,call,log.verbose,command,agent,user write = system,call,log.verbose,command,agent,user As I said, the first time I place the call from Outlook, it works fine. The trace on Asterisk shows: == Manager 'john' logged on from 192.168.2.17 -- Launching Dial(Zap/1/18005551212) on SIP/200-da5d -- Called 1/18005551212 == Manager 'john' logged off from 192.168.2.17 -- Zap/1-1 answered SIP/200-da5d -- Hungup 'Zap/1-1' Any help would be much appreciated. Rana Dutt Softel, Inc. ___ 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
[Asterisk-Users] RE: multiline IP hardphone w/ FDX speakerphone?
I use Polycom IP 500's with Asterisk. These phones have 3 line appearances and excellent full duplex speakerphones. They work very well with Asterisk, and I was able to use the Web interface to set them up quite easily. The default Web password given at voip-info.org is wrong, I added a comment on the Polycom Phones page giving the correct one (456). With Asterisk, you can do both consultative and blind transfers with this phone, and the Conference button works as you would expect. It's a very high quality product, and it has more line appearances than the cisco 7940G. I don't know whether you can get them for less than $200, I paid about $240 each. Rana Dutt -- Message: 8 Date: Mon, 6 Sep 2004 19:24:41 +0200 From: Stewart Nelson [EMAIL PROTECTED] Subject: [Asterisk-Users] multiline IP hardphone w/ FDX speakerphone? To: [EMAIL PROTECTED] Message-ID: [EMAIL PROTECTED] Content-Type: text/plain; format=flowed; charset=iso-8859-1; reply-type=original Could someone please recommend a reasonably priced IP phone that works well with *, has a decent (full duplex, echo canceling) speakerphone, has at least two line appearances, and can transfer / conference reliably? The Wiki lists 35 brands of hardphone, but: 1. Most seem to be toys. 2. For many, there is no info on e.g. speakerphone characteristics. 3. When one seems technically promising, e.g. Polycom IP500, there are *lots* of negative postings about support, integration, etc. Is there anything decent out there for $200? Thanks, Stewart ** ___ 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
[Asterisk-Users] My Cisco 7940 is not registering with Asterisk
I just got a Cisco 7940G phone running SIP firmware version POS3-06-1-00. I unlocked the phone's config, and using the soft keys, I entered the SIP Configuration menu and keyed in values for Name, Auth. Name, ProxyAddress (where I gave my Asterisk server's IP address), etc. The result is that I can make outgoing calls from this phone just fine through my Asterisk server, but I cannot call the phone. With most SIP phones, after you save the SIP settings, the phone registers with *, and you see the registration message on the * console. But in my case, saving the SIP settings had no effect, I saw no message that the phone registered. Obviously, the phone can see the * server, because outgoing calls work. Any ideas what I'm doing wrong? How do I force the 7940 to register? I have other non-Cisco phones like Snom 200 that work just fine. Here's my phone's SIP settings: Line 1 Configuration Name 201 Shortname 201 Authent. Name 201 Password none Display Name 201 Proxy Address 192.168.2.11 (the IP address of the Asterisk server) Propxy Port5060 And here's the relevant part of my sip.conf: [general] port = 5060 bindaddr = 0.0.0.0 disallow = all allow = ulaw allow = alaw [201] type=friend username=201 secret= host=dynamic context=dialout callerid=201 dtmfmode=rfc2833 mailbox=201 Thanks in advance for any help. Rana Dutt Softel, Inc Marlboro, NJ (732) 810-6707 x200 ___ 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
[Asterisk-Users] Pressing digits on SNOM phone results in letters on display
My SNOM 200 phone got into a funny mode where if I dial any digit, a letter gets displayed and sent, so dialing no longer works. For example, if I dial 9, the letter w gets displayed and sent when I press OK. How do I get it out of this mode? ___ 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
[Asterisk-Users] Call Admission Control
Let's say you have a 256 Kbps Internet connection and you're using it for voice calls. With mu-law (G.711), each call uses about 80 kbps, so you really can't have more than 3 calls active at one time. Does Asterisk support any kind of Call Admission Control where it would prevent you from originating a call if it would exceed your Internet bandwidth? For example, in this case, ideally, we would want Asterisk to present busy tone when the fourth simultaneous call is attempted. ___ 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
[Asterisk-Users] SNOM II and Siptone phone on eBay
Sorry to post this here also, but the biz list doesn't seem to have much traffic yet. I have a brand new SNOM 200 IP phone and also a new Siptone II phone available on eBay, see http://tinyurl.com/2pbng They are surplus after a customer cancelled an order. Please direct all followup questions or bids on eBay, not here. Thanks. -Ron ___ 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
RE: [Asterisk-Users] Lucent Phones
Also, check out www.citel.com This company claims to have SIP adaptors for Avaya's digital PBX phones. If they work as advertised, you can keep your Avaya/Lucent phones, throw out your legacy PBX, and connect them all to Asterisk! However, I doubt they have all the display integration working correctly. Anyone know for sure? Ron Dutt -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steven Sokol Sent: Wednesday, April 07, 2004 1:25 PM To: [EMAIL PROTECTED] Subject: RE: [Asterisk-Users] Lucent Phones -Original Message- From: [EMAIL PROTECTED] [mailto:asterisk-users- Subject: [Asterisk-Users] Lucent Phones Does Asterisk work with Lucent or any other PBX phone systems Sure. You can use Asterisk as a VoIP gateway to your existing legacy PBX. You can't plug Lucent's (Avaya's) DCP, MLX, or ATL phone sets into an Asterisk box -- the protocols are all proprietary. But you can certainly connect between the systems using analog or T1/Ei connections. Regards, Steve Steven Sokol Owner/Manager Sokol Associates, LLC Phone: 816.822.1807 IaxTel: 700.613.9004 Web:http://www.sokol-associates.com ___ 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
[Asterisk-Users] Connecting two branch offices using * and Mediatrix
Suppose a company has a U.S. office and a foreign office, and would like to make toll-free calls using IP between the offices. The U.S office will have an Asterisk system, but the foreign office has a large legacy PBX that they want to keep. One way to do this is to install a Mediatrix FXO gateway at the foreign office. This gateway could be connected to a couple of unused tip/ring station ports on the legacy PBX system. I assume that when the Mediatrix registers with Asterisk, it does so as a single SIP endpoint. Suppose this endpoint is SIP/mediatrix, and my extensions.conf routes all extensions beginning with 4 to SIP/mediatrix, like this: exten = _4XXX, Dial(SIP/mediatrix,15) When I dial such an extension from the U.S. office, presumably the Mediatrix at the foreign office will intercept the call, but then what? What I would like it do do is go off hook on one of the tip/ring ports on the legacy PBX, and dial the last 3 digits to reach another extension off the PBX. Can it be programmed to do this? Or does the caller hear the Mediatrix answering the call and then have to enter the desired extension manually? I have a similar question for the reverse direction. What exactly does the caller from the foreign office have to do to reach an extension in the U.S. office? Thanks very much. -Ron
[Asterisk-Users] Intermittent choppy speech using VoicePulse?
Yesterday evening, the speech on all the calls I made using VoicePulse sounded choppy from my side, although the called party said I sounded fine. Also, the voice mail messages I recorded calling in to the VoicePulse number sounded choppy. Calls I made over the PSTN line using my Zap interface sounded fine. This morning, though, the speech over the VoicePulse service sounds crystal clear on both sides! I haven't changed a thing in the configuration of either Asterisk or the SNOM phones I've been using. Has anyone experienced this type of intermittent choppy speech problem using VoicePulse, Asterisk, and SNOM phones? Ron attachment: winmail.dat
[Asterisk-Users] Cannot use # key to transfer calls
I cannot use the # key to transfer a call. I have two kinds of SIP phones, Grandstream and IpDialog, and the # key cannot be used to transfer on either one. If I press the # key during a call, I hear the touchtone for it, but Asterisk does nothing. The documentation for parking a call says that I must first transfer the call using #, so that's why I need this feature to work. Thanks for any pointers. -Ron Dutt ___ 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
RE: [Asterisk-Users] gs on phone ?
In your extensions.conf, the b and u are reversed. Use u${EXTEN} for
priority 2 and b${EXTEN} for priority 102.
-Ron
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Clifton
Sent: Tuesday, March 02, 2004 10:29 PM
To: [EMAIL PROTECTED]
Subject:[Asterisk-Users] gs on phone ?
I have a GS101 connected to * with sip and g729.
When an incoming call comes in from outside (via pstn for example), and no
one picks up the GS, * reports that 'the user is on the phone'. If no one
answers, I'd expect it to report 'unavailable'.
Maybe I'm not understanding the call flow ... (should it be u$ at '2', then
b$ at '102' ?) My current config for call flow seems to match others I've
seen on the wiki, etc.
my extensions.conf for the grandstream at x2015 -
[incoming]
exten = 2015,1,Dial(SIP/[EMAIL PROTECTED],20,T,t)
exten = 2015,2,Voicemail(b${EXTEN})
exten = 2015,3,Hangup
exten = 2015,102,Voicemail(u${EXTEN})
exten = 2015,103,Hangup
Thanks,
Chris Clifton
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[Asterisk-Users] Dialing out after caller leaves message
I want Asterisk to call my cell phone after someone leaves me a voice mail message. How do I do this? I cannot use Dial after the Voicemail application, e.g., [Step 1] exten = 100, 1, Dial( SIP/100, 15 ) [Step 2] exten = 100, 2, Voicemail( u100 ) [Step 3] exten = 100, 3, Dial( Zap/g1/CELL_PHONE ) because the caller will hang up after leaving the voice mail in Step 2 above, and Asterisk will terminate the script, so Step 3 will get never get executed. Is there a way to do what I want to do? -Ron attachment: winmail.dat
RE: [Asterisk-Users] Dialing out after caller leaves message
The following suggested sequence does not work: exten = 100, 1, Dial(SIP/100, 15) exten = 100, 2, Voicemail(u100) exten = h,1,Dial(Zap/g1/CELL_PHONE) The Dial command in the 3rd step will fail because the channel it uses to dial out on has hung up already! For example, if extension 101 leaves a message for ext. 100 and then hangs up, the first two steps above get executed, but after 101 hangs up, there is no current channel for Dial to use to place the call to the cell phone. Can anyone see a way to make Asterisk dial a new call on its own after the original caller has hung up? Do I need an AGI script to do this? Thanks, -Ron -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Florian Overkamp Sent: Sunday, February 29, 2004 11:47 AM To: [EMAIL PROTECTED] Subject:RE: [Asterisk-Users] Dialing out after caller leaves message Hi, -Original Message- What about putting this in a special context and using 'h'? i.e. exten = 100, 1, Dial(SIP/100, 15) exten = 100, 2, Voicemail(u100) exten = h,1,Dial(Zap/g1/CELL_PHONE) ? h will get executed on hangup. The only caveat is that if no voicemail was left, you will still get called. Is there some way to check if there are new messages and use that in h along with the Dial()? Ehm Show application HasNewVoicemail Florian ___ 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 ___ 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
RE: [Asterisk-Users] Message waiting light not coming on
The problem turned out to be in my voicemail.conf, thanks. I had the second section named [bell] instead of [default]. The MWI works perfectly now with both the Grandstream and IpDialog SIP phones. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Olle E. Johansson Sent: Thursday, February 26, 2004 4:12 AM To: [EMAIL PROTECTED] Subject:Re: [Asterisk-Users] Message waiting light not coming on Iain Stevenson wrote: Works perfectly fine for me - but I'm not using rfc2683 - my Grandstream uses the latest firmware and SIP INFO. Iain --On Thursday, February 26, 2004 12:55 am -0500 Rana Dutt [EMAIL PROTECTED] wrote: I cannot get the Message Waiting Light (MWL) on my Grandstream phone to turn on when I leave a new voice mail message for that phone. I have specified the correct mailbox in my sip.conf as follows: Have you configured mailbox account 200 in voicemail.conf ? /O ___ 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 ___ 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
RE: [Asterisk-Users] Conference and transfer
Thanks for the info. Which phones support consultation transfers? The Grandstream and IpDialog phones most certainly do not. Also, I find it disconcerting that there's a Conference button on the Grandstream phone, but when it's pressed, nothing happens. If this sends out some sort of switch-hook flash, can Asterisk intercept it, and then use the meetme app? Do the Cisco phones support conferencing using the Conference button without the need for the meetme app? -Ron -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brancaleoni Matteo Sent: Tuesday, February 24, 2004 1:09 PM To: [EMAIL PROTECTED] Subject:Re: [Asterisk-Users] Conference and transfer hi 1) Does Asterisk support a consulting transfer? E.g., call comes in, Mary answers, Mary presses Transfer and dials Joe, verifies that Joe answers and informs him who is calling, and then presses Transfer to complete the transfer? on zap channels yes on sip channels yes, depending if phone supports that too 2) How does one set up a 3-party conference? With a traditional phone system, you press the Conference button on the phone, dial the 3rd party, and press Conference again. This doesn't seem to work with Asterisk. see app_meetme Matteo -- Brancaleoni Matteo [EMAIL PROTECTED] Espia - Emmegi Srl ___ 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 ___ 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
[Asterisk-Users] Message waiting light not coming on
I cannot get the Message Waiting Light (MWL) on my Grandstream phone to turn on when I leave a new voice mail message for that phone. I have specified the correct mailbox in my sip.conf as follows: [200] type=friend username=200 host=dynamic context=dialout callerid=200 dtmfmode=rfc2833 mailbox=200 I also have an IpDialog Siptone II phone, and can't get the MWL to work on that either. Did anyone have a problem like this? -Ron ___ 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
[Asterisk-Users] Could voice mail problem be related to RAM?
I wrote to the list a couple weeks back about my voice mail messages sounding garbled. This happens no matter what phone I use to record the message, Ive tried IpDialog, Grandstream and SJ-Phone. Obviously, no one else is having this problem, since Ive never seen it discussed here. I am now wondering whether my lack of RAM may be the problem. I am running Asterisk on a Dell Dimension XPS R450 with only 128 Mb of RAM. When I run the top command, I see my RAM consumption is above 120 Mb, but I always see about 4 Mb of RAM available. Should I get more RAM? Has anyone else recorded garbled voice mail messages? -Ron
[Asterisk-Users] Conference and transfer
Two newbie questions: 1) Does Asterisk support a consulting transfer? E.g., call comes in, Mary answers, Mary presses Transfer and dials Joe, verifies that Joe answers and informs him who is calling, and then presses Transfer to complete the transfer? 2) How does one set up a 3-party conference? With a traditional phone system, you press the Conference button on the phone, dial the 3rd party, and press Conference again. This doesn't seem to work with Asterisk. Thanks, -Ron ___ 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
[Asterisk-Users] Voice mail sound distortion has everyone laughing
I wrote earlier about how garbled my voice mail messages sound, so I thought I'd attach an example of a message I recorded earlier today. Just click on the attached WAV file for a good laugh. -Original Message- From: Asterisk PBX [mailto:[EMAIL PROTECTED] Sent: Friday, February 20, 2004 10:47 PM To: Riana Dutt Subject:[PBX]: New message 1 in mailbox 201 Just wanted to let you know you were just left a 0:09 long message (number 1) in mailbox 201 from 200, on Friday, February 20, 2004 at 10:47:27 PM so you might want to check it when you get a chance. Thanks! --Asterisk msg.WAV Description: Wave audio
RE: [Asterisk-Users] softphone configs?
What does your sip.conf look like? Please include it in your next message in its entirety. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mark Messmore, Technical Support, University Telcom Inc. Sent: Wednesday, February 18, 2004 1:08 PM To: [EMAIL PROTECTED] Subject:[Asterisk-Users] softphone configs? I've tried using the x-lite softphone as well as sjphone. I've gone over my configurations a dozen times...and I always seem to get the following error: Feb 18 11:30:16 NOTICE[1125329600]: chan_sip.c:5577 handle_request: Registration from 'Mark sip:[EMAIL PROTECTED]' failed for '192.168.5.64' FYI...I'm trying to do all my voip internally, nothing to the outside world yet. If anyone could give me an idea I'd appreciate it. Thanks. Mark ___ 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 ___ 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
[Asterisk-Users] Slow, distorted speech on voice mail messages
When I play back my voice mail messages, the words sound very s-l-o-w-e-d down and distorted. It's like I'm speaking in slow motion. I've tried recording using an IpDialog SipTone phone, a Grandstream phone and SJ-Phone, and the problem always happens. So it's not the phone. When I talk from one phone to another, the speech sounds normal. The problem is only happening for recorded voice mail messages. All of my different types of SIP phones are using ulaw codecs and my sip.conf contains disallow=all followed by allow=ulaw for all phones. Asterisk is a CVS build from 2/1/04. Any insights would be much appreciated. -Ron ___ 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
[Asterisk-Users] Double digits seen using Grandstream phones
My attempts to use voice mail from my Grandstream Budgetone 101 phone always fail because Asterisk is seeing either double digits or dropped digits, no matter what dtmfmode setting I try. Here is what happens for each mode: dtmfmode=info, phone set to send INFO 101: every digit is seen double, e.g. 123 is seen as 112233 dtmfmode=rfc2833, phone set to RFC2833: some digits are doubled in random places, e.g., 123 is sometimes seen as 1123, sometimes as 1233 dtmfmode=inband, phone set to send in-audio: some digits are dropped, e.g., 123 becomes 23 or 12 I have upgraded the firmware in my phones to 1.0.4.46 and Asterisk is a CVS from 2/1/04. The codec in the phones is set to PCMU (ulaw) as first preference, and the sip.conf has disallow=all followed by allow=ulaw for each phone. Early dial is turned off. Can anyone help me? Thanks, -Ron attachment: winmail.dat
RE: [Asterisk-Users] Re: Double digits seen using Grandstream phones
Yes!! Applying the patch and using dtmfmode=rfc2833 cleared up the problem. The only side effect I see now are some console warnings RTP Read error: Resource temporarily unavailable, but they seem harmless, and the system is working fine. Thank you! -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of James H. Cloos Jr. Sent: Tuesday, February 17, 2004 11:51 AM To: [EMAIL PROTECTED] Subject:[Asterisk-Users] Re: Double digits seen using Grandstream phones Rana == Rana Dutt [EMAIL PROTECTED] writes: Rana My attempts to use voice mail from my Grandstream Budgetone 101 Rana phone always fail because Asterisk is seeing either double Rana digits or dropped digits, no matter what dtmfmode setting I try. Try the patch in bug number 1034: http://bugs.digium.com/bug_view_advanced_page.php?bug_id=0001034 That deals with doubles due to packet reording in transit and also due to phones that interleave voice packets with 2833 packets. After applying that patch, use rfc2833 for dtmf. -JimC ___ 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 ___ 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
RE: [Asterisk-Users] Speech between Grandstream phones sounds like talking under water
I was able to solve the audio quality problem by going to www.grandstream.com/BETATEST and downloading the latest beta firmware, version 1.0.4.46. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Philipp von Klitzing Sent: Tuesday, February 17, 2004 7:14 AM To: [EMAIL PROTECTED] Subject:Re: [Asterisk-Users] Speech between Grandstream phones sounds like talking under water Hi! You need to add this to EACH and EVERY sip user, not just in [general]: disallow=all allow=ulaw allow=alaw See also: http://www.voip-info.org/wiki-Asterisk+phone+grandstream+budgetone Cheers, Philipp [200] type=friend username=200 host=dynamic context=home reinvite=no canreinvite=no [201] type=friend username=201 host=dynamic context=home reinvite=no canreinvite=no I turned on sip debug, and noticed the following in the output: v=0 s=SIP Call c= IN IP4 192.168.2.29 m= audio 5004 RTP/AVP 0 a=rptmap:0 PCMU/8000 a=ptime:20 Found audio format UNKN Found description format PCMU Capabilities: us - 4, them 4/0, combined - 4 Non-codec capabilities: us - 1, them - 0, combined 0 Does anyone know why this could be happening? Thanks, Ron ___ 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 ___ 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
RE: [Asterisk-Users] Buzzing on Grandstream phones
Try downloading the latest firmware, version 1.0.4.46 from www.grandstream.com/BETATEST I used to have bad audio problems on my Budgetone 101's until I upgraded their firmware. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Todd Wallace Sent: Tuesday, February 17, 2004 4:04 PM To: [EMAIL PROTECTED] Subject:[Asterisk-Users] Buzzing on Grandstream phones I get a low buzzing noise on my Grandstream phones when placing calls. Any one know how to get rid of that... Todd ___ 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 ___ 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
RE: [Asterisk-Users] Howto apply a patch, diff file
Type the command patch file.c fix.diff where fix.diff is the patch to apply to file.c. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jan Larsen Sent: Tuesday, February 17, 2004 9:03 PM To: [EMAIL PROTECTED] Subject:[Asterisk-Users] Howto apply a patch, diff file Hi all I am new to linux and * , so could someone please explain how to apply a patch file (diff extension) Kind regards Jan ___ 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 ___ 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
[Asterisk-Users] Speech between Grandstream phones sounds like talking under water
When I make a simple phone call from one Budgetone 101 to another, the speech sounds slurred and slow, sort of like the person is talking under water. Both phones and the Asterisk server are on the same subnet. Both phones are configured to use the PCMU (ulaw) codec as first choice, and the Voice Frames per TX parameter is set to 2. Incidentally, if I directly IP dial from one phone to the other (bypassing Asterisk) the speech sounds excellent. I'm running a CVS build from Feb. 1, 2004, and there is a Digium X100P card with one incoming CO line in my machine. The first part of my sip.conf looks like this: [general] port=5060 binaddr=0.0.0.0 disallow=all allow=ulaw [200] type=friend username=200 host=dynamic context=home reinvite=no canreinvite=no [201] type=friend username=201 host=dynamic context=home reinvite=no canreinvite=no I turned on sip debug, and noticed the following in the output: v=0 s=SIP Call c= IN IP4 192.168.2.29 m= audio 5004 RTP/AVP 0 a=rptmap:0 PCMU/8000 a=ptime:20 Found audio format UNKN Found description format PCMU Capabilities: us - 4, them 4/0, combined - 4 Non-codec capabilities: us - 1, them - 0, combined 0 Does anyone know why this could be happening? Thanks, Ron attachment: winmail.dat
