[asterisk-users] Type 102 Millwatt Test Line
Does anybody know a type 102 milliwatt test number that I can dial in the USA? I need this in order to configure my rxgain and txgain. My analog line provider, ATT Repair Center was so confuse, when called them. Thanks in advance. -John _ Stay up to date on your PC, the Web, and your mobile phone with Windows Live http://clk.atdmt.com/MRT/go/119462413/direct/01/___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF issue
In my setup, I am using TDM Wildcards analog connections and the Asterisk PBX
box does the converting to my SIP Phones. I had similar problem, when Asterisk
could not recognize my DTMF tones, so I had to tune the FXO modules. Here is
the link to the page:
http://www.voip-info.org/wiki/view/Asterisk+fxotune
If you are you using pure SIP Protocal, you may want to ask your SIP provider
for the suggested dtmfmode, even though RFC2833 is recommended by most. I did
have the same problems with this in the past, when I was testing with SIP
providers and I never solved it. Therefore, I went with the TDM Wildcard route
with analog lines. Good luck
Date: Thu, 16 Oct 2008 13:28:54 +0300
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] DTMF issue
Dear All,
I have the following scenario:
My customer dial a DID number and it'll be forwarded to my asterisk server by
the below trunk defined in sip.conf:
[sip_proxy1]
type=peer
context=stations
host=81.201.82.112
disallow=all
allow=g729
allow=alaw
allow=ulaw
dtmfmode=RFC2833
relaxdtmf=yes
canreinvite=no
The above trunk will use the context stations defined in extensions.conf as
follow:
[stations]
exten = _X.,1,Gotoif($[${EXTEN} = 112] ? 21)
exten = _X.,2,DeadAGI,a2billing.php|3
exten = _X.,3,Wait,2
exten = _X.,4,Hangup
exten = _X.,21,AGI,a2billing.php|3
exten = _X.,22,Hangup
The System will ask the user to enter his PIN number...The problem is that
sometime the system recognize the PIN entered and sometimes the PIN is not
recognized...I'm using RFC2833 as dmf mode...
I would like to know if my config is correct or I need o add something to it or
there is a BUG on asterisk server regarding DTMF?
Please note that I'm using Asterisk 1.4.21.2
Regards
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Re: [asterisk-users] BLF Compatible Phones
I have the SPA962 with the SPA932 side car and it works great, once I got it to configure correctly for my Asterisk PBX system, which has a Digium TDM03B WILD CARD installed. I like the Cisco/Linksys SPA models because they have good and easy configuration webpages for the VOIP phones. Date: Mon, 19 May 2008 18:51:09 +0200 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] BLF Compatible Phones Hmm, i dont like aastra really much, their transfer management is not human friendly ;) Polycom or Snom is our choice (snom 340) 2008/5/18 Sigma Networks [EMAIL PROTECTED]: [EMAIL PROTECTED] wrote: I am new to asterisk and am looking to setup a small office with 4-6 IP phones and 4 analog lines from the local telco (primary line with HUNT to the other lines). I am considering purchase of a Digium AEX800. One of the features that will be important (particularly for the receptionist desk is to show status of the other lines in use). I don't want the receptionist to pick up a line if it being used. Is there a list of phones that are BLF (Busy Lamp Field) compatible? I'm assuming (after reading tons of misc articles) that this is what I need in order for the receptionist not to pick up lines in use. If this is not the case please set me straight. I am considering the cisco 7960's, linksys SPA942, and possibly some polycom phones. I was leaning toward the 7960 but I've read that it is not BLF compatible. Are there any workarounds for this? I am new to the game and would be grateful for any recommendations on which phones would be the easiest to setup, etc. I currently have a working asterisk install at home with a single cisco 7960 registered which isn't hooked up to any trunks as of yet. Thanks, Dayton Gray ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users I concur with one of the earlier posts that the Aastra 57i (or 57iCT) with the 560m LCD side car works very well. LCD side cars are ideal because they're alway up to date and paper based sidecars look messy. The Aastra 57iCT is unique in that it adds a portable handset which gives the admin the ability to go to other parts of the office and still handle calls. The Polycom 650 with it's sidecar also works well with BLF. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ E-mail for the greater good. Join the i’m Initiative from Microsoft. http://im.live.com/Messenger/IM/Join/Default.aspx?source=EML_WL_ GreaterGood___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF issues in 1.4.19 with missing digits
I had a similar problem when using the TDM03B card with 3 fxo module. In my cas,e the issue stemmed from a noisy analog line from ATT, so I had to tune my TDM card by using fxotune utility. I hope this helps. Check this link out: http://www.voip-info.org/wiki/view/Asterisk+fxotune -John Date: Fri, 2 May 2008 15:14:54 -0500 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF issues in 1.4.19 with missing digits Hello, all! Trying to figure out an issue with DTMF recognition with 1.4.19. This is somewhat similar to the issue described here: http://bugs.digium.com/view.php?id=11740, but it might be a different issue altogether. I have 1.4.19 running with TE212P on a US PRI. I'm sending digits 82322. Sometimes the digits are making it all in the asterisk, and sometimes some are missing. In the case when the digits are all caught, my DTMF log enteries are something like this: snip [May 2 14:48:56] DTMF[28649]: channel.c:2092 __ast_read: DTMF end '8' received on Zap/1-1, duration 0 ms [May 2 14:48:56] DTMF[28649]: channel.c:2144 __ast_read: DTMF end accepted without begin '8' on Zap/1-1 [May 2 14:48:56] DTMF[28649]: channel.c:2155 __ast_read: DTMF end passthrough '8' on Zap/1-1 [May 2 14:48:57] DTMF[28649]: channel.c:2092 __ast_read: DTMF end '2' received on Zap/1-1, duration 0 ms [May 2 14:48:57] DTMF[28649]: channel.c:2144 __ast_read: DTMF end accepted without begin '2' on Zap/1-1 [May 2 14:48:57] DTMF[28649]: channel.c:2155 __ast_read: DTMF end passthrough '2' on Zap/1-1 [May 2 14:48:57] DTMF[28649]: channel.c:2092 __ast_read: DTMF end '3' received on Zap/1-1, duration 0 ms [May 2 14:48:57] DTMF[28649]: channel.c:2144 __ast_read: DTMF end accepted without begin '3' on Zap/1-1 [May 2 14:48:57] DTMF[28649]: channel.c:2155 __ast_read: DTMF end passthrough '3' on Zap/1-1 [May 2 14:48:58] DTMF[28649]: channel.c:2092 __ast_read: DTMF end '2' received on Zap/1-1, duration 0 ms [May 2 14:48:58] DTMF[28649]: channel.c:2144 __ast_read: DTMF end accepted without begin '2' on Zap/1-1 [May 2 14:48:58] DTMF[28649]: channel.c:2155 __ast_read: DTMF end passthrough '2' on Zap/1-1 [May 2 14:48:58] DTMF[28649]: channel.c:2092 __ast_read: DTMF end '2' received on Zap/1-1, duration 0 ms [May 2 14:48:58] DTMF[28649]: channel.c:2144 __ast_read: DTMF end accepted without begin '2' on Zap/1-1 [May 2 14:48:58] DTMF[28649]: channel.c:2155 __ast_read: DTMF end passthrough '2' on Zap/1-1 [May 2 14:48:59] DTMF[28649]: channel.c:2092 __ast_read: DTMF end '9' received on Zap/1-1, duration 0 ms [May 2 14:48:59] DTMF[28649]: channel.c:2144 __ast_read: DTMF end accepted without begin '9' on Zap/1-1 [May 2 14:48:59] DTMF[28649]: channel.c:2155 __ast_read: DTMF end passthrough '9' on Zap/1-1 [May 2 14:49:00] DTMF[28649]: channel.c:2092 __ast_read: DTMF end '9' received on Zap/1-1, duration 0 ms [May 2 14:49:00] DTMF[28649]: channel.c:2144 __ast_read: DTMF end accepted without begin '9' on Zap/1-1 [May 2 14:49:00] DTMF[28649]: channel.c:2155 __ast_read: DTMF end passthrough '9' on Zap/1-1 [May 2 14:49:00] DTMF[28649]: channel.c:2092 __ast_read: DTMF end '9' received on Zap/1-1, duration 0 ms [May 2 14:49:00] DTMF[28649]: channel.c:2144 __ast_read: DTMF end accepted without begin '9' on Zap/1-1 [May 2 14:49:00] DTMF[28649]: channel.c:2155 __ast_read: DTMF end passthrough '9' on Zap/1-1 [May 2 14:49:01] DTMF[28649]: channel.c:2092 __ast_read: DTMF end '9' received on Zap/1-1, duration 0 ms [May 2 14:49:01] DTMF[28649]: channel.c:2144 __ast_read: DTMF end accepted without begin '9' on Zap/1-1 [May 2 14:49:01] DTMF[28649]: channel.c:2155 __ast_read: DTMF end passthrough '9' on Zap/1-1 /snip In the case when digits are not fully recognized (one is missing), I get this: snip [May 2 14:36:16] DTMF[28461]: channel.c:2092 __ast_read: DTMF end '8' received on Zap/1-1, duration 0 ms [May 2 14:36:16] DTMF[28461]: channel.c:2128 __ast_read: DTMF begin emulation of '8' with duration 100 queued on Zap/1-1 [May 2 14:36:16] DTMF[28461]: channel.c:2092 __ast_read: DTMF end '2' received on Zap/1-1, duration 0 ms [May 2 14:36:16] DTMF[28461]: channel.c:2098 __ast_read: DTMF end '2' put into dtmf queue on Zap/1-1 [May 2 14:36:16] DTMF[28461]: channel.c:2237 __ast_read: DTMF end emulation of '8' queued on Zap/1-1 [May 2 14:36:16] DTMF[28461]: channel.c:1961 __ast_read: DTMF begin emulation of '2' with duration 100 queued on Zap/1-1 [May 2 14:36:16] DTMF[28461]: channel.c:2092 __ast_read: DTMF end '3' received on Zap/1-1, duration 0 ms [May 2 14:36:16] DTMF[28461]: channel.c:2144 __ast_read: DTMF end accepted without begin '3' on Zap/1-1 [May 2 14:36:16] DTMF[28461]: channel.c:2155 __ast_read: DTMF end passthrough '3' on Zap/1-1 [May 2 14:36:17] DTMF[28461]: channel.c:2237 __ast_read: DTMF end
[asterisk-users] Callerid Error
Asterisk Users, I am running a Debian Etch system with Asterisk 1.4.11 with a TDM03B card. Once in awhile, I get this error on the Asterisk, which causes my channels to be busy/congested, leaving me with just one channel to recieve and make calls: NOTICE[31454]: chan_zap.c:6367 ss_thread: Got event 17 (Polarity Reversal)... WARNING[31454]: chan_zap.c:6499 ss_thread: CallerID returned with error on channel 'Zap/3-1' What could be causing this issue? Any would input would be greatly appreciated. Thanks In Advance, John _ Use video conversation to talk face-to-face with Windows Live Messenger. http://www.windowslive.com/messenger/connect_your_way.html?ocid=TXT_TAGLM_WL_Refresh_messenger_video_042008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] problem TDM01B
I have a TDM04B card and not seen this issue. You may need to check your dial plan. Date: Sat, 12 Apr 2008 14:00:49 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] problem TDM01B hI list, I have some problems with a TDM01B , when I am talking on the phone with another person it cuts himself the call, this alone I am presented when I make calls to the pstn, with internal extensions I don't have problems I show them the log in the CLI -- Nobody picked up in 68000 ms -- Hungup 'Zap/4-1' -- Executing [EMAIL PROTECTED]:2] Hangup(SIP/113-081cf588, ) in new stack == Spawn extension (cyber, 2768000, 2) exited non-zero on 'SIP/113-081cf588' Some person of the list that has presented the same problem with this card, and it finds it solved greetings rickygm ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Get in touch in an instant. Get Windows Live Messenger now. http://www.windowslive.com/messenger/overview.html?ocid=TXT_TAGLM_WL_Refresh_getintouch_042008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Low Volume on Recorded Voicemail Messages
Asterisk Users, I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a Debian Etch system. On the recorded voice mail messages, the volume is really low when retrieving them with my cell phone. I tried with multiple cell phones with the volume level high and still, the same problem. I tried everything from changing rxgain and txgain in the zapata.conf to added the g option in the voicemail option. So, I tried writing a script with sox to increase the volume of the voicemail message in the INBOX, but did not have any luck Can somebody help me with this? Or have a patch for the voicemail application that would work to increase the volume of voicemail messages. Any input would be greatly appreciated. Thanks in advance. Best Regards, John _ More immediate than e-mail? Get instant access with Windows Live Messenger. http://www.windowslive.com/messenger/overview.html?ocid=TXT_TAGLM_WL_Refresh_instantaccess_042008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail- Recorded Mesage Low Volume
Thanks. I will give this a try. -John From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Date: Wed, 2 Apr 2008 09:29:48 -0700 Subject: Re: [asterisk-users] Voicemail- Recorded Mesage Low Volume On Apr 1, 2008, at 5:22 PM, [EMAIL PROTECTED] wrote: Can the volume of the recorded voice mail message be changed? If so, what I am doing wrong? Any input would be greatly appreciated. Thanks. I had a similar problem in our setup where we e-mail the recorded messages to e-mail retrieval. But this also helps standard phone retrieval too. What I did was edit the /usr/sbin/safe_asterisk script and add: PATH=/usr/local/bin:$PATH At the top of the script. This would let me override the default sox implementation that Asterisk uses. Then I loaded in a script (called sox) that would compress and normalize the recorded audio (It compresses to deal with the spikes of the noise of the handset being hung up, etc.). It works pretty well for us and makes the volume pretty good so we don't have to crank up the volume on our computers or phones to listen to voicemail messages. And we can't adjust the rxgain as it is already a good volume for normal calls. Daniel --CUT-- #!/bin/sh # # $1 = -v # $2 = number # $3 = inFile # $4 = outFile # REALSOX=/usr/bin/sox if [ $1 != -v ]; then $REALSOX $* exit $? fi INFILE=$3 OUTFILE=$4 # # Perform the gain adjustment. # $REALSOX $INFILE $OUTFILE compand 0.1,0.3 -60,-60,-30,-15,-20,-12,-4,-8,-2,-7 0 0 0.2 --CUT-- ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ More immediate than e-mail? Get instant access with Windows Live Messenger. http://www.windowslive.com/messenger/overview.html?ocid=TXT_TAGLM_WL_Refresh_instantaccess_042008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail- Recorded Mesage Low Volume
Asterisk Users,
I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a Debian
Etch system. On the recorded voice mail messages, the volume is really low
when retrieving them with my cell phone. I tried with multiple cell phones
with the volume level high and still, the same problem. I tried to increase the
rxgain to 12.2 in the zapata.conf file and it had no affect on the volume. So,
tried to add the g option on the asterisk voicemail command, like this:
exten = s-NOANSWER,1, Voicemail(u${EXTENSION}|g(10))
and the volume remained the same. I cannot find any good suggestions on the
internet in dealing with this issue.
Can the volume of the recorded voice mail message be changed? If so, what I
am doing wrong? Any input would be greatly appreciated. Thanks.
Best Regards,
John
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Re: [asterisk-users] Spam: Voicemail- Recorded Mesage Low Volume
When people leave me messages, both on the cellphone and POTS phones, on the
recorded Asterisk voicemail message volume is really low. I could barely hear
my voicemail messages, when retrieving them, either cellphone or POTS line.
The voice mail prompts and sound recordings are fine, but the problem lies when
people leave voicemails on the Asterisk box. I could barely hear my voicemail
messages.
What can cause this?
Best Regards,
John
Date: Tue, 1 Apr 2008 14:46:40 -0500
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Spam: Voicemail- Recorded Mesage Low Volume
Do you see the
same volume issues dialing in with a ‘normal’ POTS phone?
Are these the
standard recordings that come with Asterisk or some custom recordings?
…brig
Brig C. McCoy
ThyssenKrupp
Access Corp
Network
Administrator
Grandview, MO 64030
816-767-5577
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan
Sent: Tuesday, April 01, 2008 2:32
PM
To: Asterisk Users
Subject: Spam:[asterisk-users]
Voicemail- Recorded Mesage Low Volume
Asterisk Users,
I am running Asterisk 1.4.11, Zaptel 1.4.5.1, and Librpi 1.4.1 on a
Debian Etch system. On the recorded voice mail messages, the
volume is really low when retrieving them with my cell phone. I tried
with multiple cell phones with the volume level high and still, the same
problem. I tried to increase the rxgain to 12.2 in the zapata.conf file and it
had no affect on the volume. So, tried to add the g option on
the asterisk voicemail command, like this:
exten = s-NOANSWER,1, Voicemail(u${EXTENSION}|g(10))
and the volume remained the same. I cannot find any good suggestions on
the internet in dealing with this issue.
Can the volume of the recorded voice mail message be changed? If
so, what I am doing wrong? Any input would be greatly appreciated.
Thanks.
Best Regards,
John
Use video conversation to talk face-to-face with Windows
Live Messenger. Get started!
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Re: [asterisk-users] SPA-962+ SPA-932- blf function
On the Asterisk CLI show hints Registered Asterisk Dial Plan Hints =- [EMAIL PROTECTED]: SIP/211 State:IdleWatchers 1 - 1 hints registered On the Asterisk CLI sip show subcriptions Peer UserCall IDExtensionLast state TypeMailbox x.x.x.x 218 ad7e0925-24 [EMAIL PROTECTED] Idle dialog-info+xml none 1 active SIP subscription I do have real ip address for my asterisk server under the Peer column. This is the output I get on the Asterisk CLI , when I am in a call with extension 211 (SPA-941). So on my SPA-962 + SPA-932, the LED state remains GREEN, because Asterisk thinks it is in Idle state, which extension 211 is clearly not. Why is that? Best Regards, John Date: Fri, 28 Mar 2008 16:53:35 +1100 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SPA-962+ SPA-932- blf function We have BLF buttons working fine on the SPA932 side-car. What does show hints tell you under Asterisk, and what syntax did you use when configuring the side-car buttons? John Meksavan wrote: Asterisk Users, I am running Asterisk 1.4.11 on Debian Etch system with the TDM03B wildcard. I recently purchased a SPA-962 and SPA-932- the sidecar for our receptionist. After reading many forum postings on how to configure the side car, I uprgraded the SPA-962 software to 5.1.18(SC) version. I got the sidecar to subscribed to an extension on the Asterisk server, but the LED state on the SPA-932 never changes even whenI am a call with that extension on another VOIP phone- SPA-941. I got the speed dial function to work, but the blf function does not appear to work. Did anybody get the blf function to work? What I am doing wrong? Any input would be greatly appreciated. Thanks in advance. Regards, John How well do you know your celebrity gossip? Talk celebrity smackdowns here. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Windows Live Hotmail is giving away Zunes. http://www.windowslive-hotmail.com/ZuneADay/?locale=en-USocid=TXT_TAGLM_Mobile_Zune_V3___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SPA-962+ SPA-932- blf function
Thanks for you guys help. The status LED on the sidecar takes an awfully look time to change from GREEN to RED and vice versa. Some times, it would reguire up to 15-20 minutes at beginning or ending the call on the extension. What would cause the delay? Is it my network? Best Regards, John Date: Fri, 28 Mar 2008 09:05:02 -0600 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] SPA-962+ SPA-932- blf function John Meksavan wrote: Asterisk Users, I am running Asterisk 1.4.11 on Debian Etch system with the TDM03B wildcard. I recently purchased a SPA-962 and SPA-932- the sidecar for our receptionist. After reading many forum postings on how to configure the side car, I uprgraded the SPA-962 software to 5.1.18(SC) version. I got the sidecar to subscribed to an extension on the Asterisk server, but the LED state on the SPA-932 never changes even when I am a call with that extension on another VOIP phone- SPA-941. I got the speed dial function to work, but the blf function does not appear to work. Did anybody get the blf function to work? What I am doing wrong? Any input would be greatly appreciated. Thanks in advance. Regards, John How well do you know your celebrity gossip? Talk celebrity smackdowns here. http://originals.msn.com/thebigdebate?ocid=T002MSN03N0707A ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users To make it work properly I had to add the following to sip.conf: allowsubscribe=yes notifyringing=yes limitonpeer=yes notifyhold=yes See if that helps. -Sean ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ How well do you know your celebrity gossip? http://originals.msn.com/thebigdebate?ocid=T002MSN03N0707A___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SPA-962+ SPA-932- blf function
Asterisk Users, I am running Asterisk 1.4.11 on Debian Etch system with the TDM03B wildcard. I recently purchased a SPA-962 and SPA-932- the sidecar for our receptionist. After reading many forum postings on how to configure the side car, I uprgraded the SPA-962 software to 5.1.18(SC) version. I got the sidecar to subscribed to an extension on the Asterisk server, but the LED state on the SPA-932 never changes even when I am a call with that extension on another VOIP phone- SPA-941. I got the speed dial function to work, but the blf function does not appear to work. Did anybody get the blf function to work? What I am doing wrong? Any input would be greatly appreciated. Thanks in advance. Regards, John _ How well do you know your celebrity gossip? http://originals.msn.com/thebigdebate?ocid=T002MSN03N0707A___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Callerid Error- Causing All Zap Channels Busy
Asterisk Users, I am running Asterisk-1.4.11 on a Debian Etch system. On an occasion, when customer calls into my Asterisk Box, I get this error messagefrom Asterisk CallerID returned with error on channel Zap/3-1 , causing all my zap channels to be busy. So, I cannot make any calls in, nor out. I am located in the United States. Is there any other suggestions, besides adding busydetect= yes and busycount=8? Any other suggestions would be appreciated. Thanks in advance. Here is what my zapata.conf looks like: [channels] ;context=telewest_pstn context=default switchtype=national signalling=fxs_ks rxwink=300 ; Atlas seems to use long (250ms) winks channel = 2-4 adsi=yes usecallerid=yes cidsignalling=bell callerid=asreceived hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes rxgain=00.0 txgain=00.0 group=0 callgroup=0 pickupgroup=0 busydetect=yes busycount=8 echotraining=yes immediate=yes relaxdtmf=yes Best Regards, John _ Connect and share in new ways with Windows Live. http://www.windowslive.com/share.html?ocid=TXT_TAGHM_Wave2_sharelife_012008___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Unusual DTMF behavior
What is your setup, hardware wise? If you have the digium cards- FXO or FXS, you must make sure you tune them. I had issues with DTMF's, when I went live with my Asterisk system. Once I tune them, everything worked great. Date: Wed, 24 Oct 2007 09:05:35 -0500 From: [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Subject: [asterisk-users] Unusual DTMF behavior We are having an issue where DTMF is not being sent out right away and the tone duration is inconsistent. For a test we send a '5', then a second later we send a '9', and then five seconds later we send a '5'. If you look at the logs below you can see the first '5' is played right away, then the '9' comes in and gets queued, but it doesn't start playing until five seconds later and it plays for six seconds. Then the last '5' is played. The DTMF is coming in as only 'end' packets and we can't change that. For this reason we have turned on rfc2833compensate. Using Asterisk 1.4.11. Any ideas? asteriskpri04*CLI Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 499/0x1F3) (Terminator) Message type: CONNECT (7) q931.c:3371 q931_receive: call 33267 on channel 3 enters state 10 (Active) Protocol Discriminator: Q.931 (8) len=5 Call Ref: len= 2 (reference 499/0x1F3) (Originator) Message type: CONNECT ACKNOWLEDGE (15) [Oct 23 10:39:56] DEBUG[6136]: chan_zap.c:1413 zt_enable_ec: Echo cancellation already on -- Zap/3-1 answered SIP/test.com-08dc1ef8 [Oct 23 10:39:58] DTMF[13914]: channel.c:2346 __ast_read: DTMF end '5' received on SIP/test.com-08dc1ef8, duration 0 ms [Oct 23 10:39:58] DTMF[13914]: channel.c :2382 __ast_read: DTMF begin emulation of '5' with duration 100 queued on SIP/test.com-08dc1ef8 [Oct 23 10:39:58] DEBUG[13914]: chan_zap.c:1048 zt_digit_begin: Started VLDTMF digit '5' [Oct 23 10:39:59] DTMF[13914]: channel.c:2346 __ast_read: DTMF end '9' received on SIP/test.com-08dc1ef8, duration 0 ms [Oct 23 10:39:59] DTMF[13914]: channel.c:2352 __ast_read: DTMF end '9' put into dtmf queue on SIP/test.com-08dc1ef8 [Oct 23 10:39:59] DTMF[13914]: channel.c:2434 __ast_read: DTMF end emulation of '5' queued on SIP/test.com-08dc1ef8 [Oct 23 10:39:59] DEBUG[13914]: chan_zap.c:1083 zt_digit_end: Ending VLDTMF digit '5' [Oct 23 10:40:04] DTMF[13914]: channel.c:2215 __ast_read: DTMF begin emulation of '9' with duration 100 queued on SIP/test.com-08dc1ef8 [Oct 23 10:40:04] DEBUG[13914]: chan_zap.c:1048 zt_digit_begin: Started VLDTMF digit '9' [Oct 23 10:40:04] DTMF[13914]: channel.c:2346 __ast_read: DTMF end '5' received on SIP/test.com-08dc1ef8, duration 0 ms [Oct 23 10:40:04] DTMF[13914]: channel.c:2352 __ast_read: DTMF end '5' put into dtmf queue on SIP/test.com-08dc1ef8 [Oct 23 10:40:10] DTMF[13914]: channel.c:2465 __ast_read: DTMF end emulation of '9' queued on SIP/test.com-08dc1ef8 [Oct 23 10:40:10] DEBUG[13914]: chan_zap.c:1083 zt_digit_end: Ending VLDTMF digit '9' [Oct 23 10:40:10] DTMF[13914]: channel.c:2215 __ast_read: DTMF begin emulation of '5' with duration 100 queued on SIP/test.com-08dc1ef8 [Oct 23 10:40:10] DEBUG[13914]: chan_zap.c:1048 zt_digit_begin: Started VLDTMF digit '5' [Oct 23 10:40:10] DTMF[13914]: channel.c:2465 __ast_read: DTMF end emulation of '5' queued on SIP/test.com-08dc1ef8 [Oct 23 10:40:10] DEBUG[13914]: chan_zap.c:1083 zt_digit_end: Ending VLDTMF digit '5' Thanks, Jason _ Peek-a-boo FREE Tricks Treats for You! http://www.reallivemoms.com?ocid=TXT_TAGHMloc=us___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype + Asterisk
Alejandro, Thanks for replying. I did come by this website before. I was just wandering, if anybody actually tried Skype with Asterisk. My experimentation with the Sip Protocol and Asterisk is at end because I could never get QOS with any sip provider, ie Broadvoice, Vitelity, and Teliax, when connecting directly to the General Internet. In my past experience, Skype has been the only VOIP that works very well. If I could just integrate this with my Asterisk at work, it would really make my boss happy. From: Alejandro Lengua [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Skype + Asterisk Date: Fri, 14 Sep 2007 13:02:19 -0500 Did you got a response for your questions? Recently found this URL in Google SiSky http://www.yeastar.com/ProductsforAsterisk.asp Regards, Alejandro Lengua On 9/6/07, John Meksavan [EMAIL PROTECTED] wrote: Has anybody ever integrated Skype with Asterisk? If you have, which software would you recommend to accomplish such a task? ChanSkype? And how reliable are the calls? Did the DTMF tones work? Thanks in advance. ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Get the device you want, with the Hotmail® you love. http://www.microsoft.com/windowsmobile/mobilehotmail/default.mspx?WT.mc_ID=MobileHMTagline ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] CallWithUs Service?
Asterisk Users, I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? Thanks in advance. -John _ Gear up for Halo® 3 with free downloads and an exclusive offer. http://gethalo3gear.com?ocid=SeptemberWLHalo3_MSNHMTxt_1 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] CallWithUs Service?
Ira and Doug, Thanks for your inputs. It seems like there are so many mixed reviews on every sip provider. In the past, I have used Broadvoice, Vitelity, and Teliax. All three have all the same problems- call quality and DTMF Tones. Some days, it would work perfectly fine, while on other days you wished you never went the sip route. There has to be some reasonable priced sip provider that would perform just as well as ATT. Does it exist? -John From: Ira [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] CallWithUs Service? Date: Thu, 13 Sep 2007 14:44:08 -0700 At 12:32 PM 9/13/2007, you wrote: I am thinking about selecting CALLWITHUS as my sip provider. Has anybody ever used them? How was the call quality? DTMF Tones issues? I use them as a a backup and they seem fine. Ira ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Discover sweet stuff waiting for you at the Messenger Cafe. Claim your treat today! http://www.cafemessenger.com/info/info_sweetstuff.html?ocid=TXT_TAGHM_SeptHMtagline2 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skype + Asterisk
Has anybody ever integrated Skype with Asterisk? If you have, which software would you recommend to accomplish such a task? ChanSkype? And how reliable are the calls? Did the DTMF tones work? Thanks in advance. _ Discover sweet stuff waiting for you at the Messenger Cafe. Claim your treat today! http://www.cafemessenger.com/info/info_sweetstuff.html?ocid=TXT_TAGHM_SeptHMtagline2 ___ Sign up now for AstriCon 2007! September 25-28th. http://www.astricon.net/ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Password Issue
The commas do work also. Thanks again, Moj. From: Mojo with Horan Company, LLC [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail Password Issue Date: Tue, 28 Aug 2007 14:53:26 -0800 John, glad it worked for you. Since you didn't feel you needed a name or email address, you might as well try just commas as delimiters: 201 = 1234,, Moj John Meksavan wrote: Mojo, Thanks for helping me with this issue. You must have a NAME and EMAIL address after putting in the voicemail pin. I just migrate to Asterisk 1.4.x from 1.2.13, so I am still trying to get use to all the new stuff in the newer version. In Asterisk 1.2.13, it is not necessary to have a name and email address. Thanks again for your help in resolving this issue. Best Regards, John From: Mojo with Horan Company, LLC [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail Password Issue Date: Tue, 28 Aug 2007 14:10:19 -0800 While I can't say this won't work the way you have it, I CAN say it's not the way mine is set up and it's not a way I've SEEN it ever set up. Could it just be complaining that you've got nothing on the right side of the = for mailbox 200? Or could it be complaining that you don't have anything past the pin number on the other lines? Try: 201 = 1234,Name or 201 = 1234,Name,email I'm thinking it's my first suggestion, though. To test that, try adding another without a pin number: 199 = and see if you then get two of the variable has bad format error messages Moj John Meksavan wrote: Here is my voicemail.conf file: [default] 200 = 201 = 1234 225 = 1234 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ More photos, more messages, more storageget 2GB with Windows Live Hotmail. http://imagine-windowslive.com/hotmail/?locale=en-usocid=TXT_TAGHM_migration_HM_mini_2G_0507 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Now you can see trouble before he arrives http://newlivehotmail.com/?ocid=TXT_TAGHM_migration_HM_viral_protection_0507 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail Password Issue
Asterisk Users, I am running Asterisk-1.4.11 with Zaptel-1.4.4 on Debian Etch System 2.9.18-4-amd64. A TDM03B is installed on the Debian System. Every time, I try to change my voicemail pin via the Sip phone, the voicemail.conf does not get modify and I see this warning message on the Asterisk command line: [Aug 29 00:12:23] WARNING[19142]: app_voicemail.c:799 vm_change_password: variable has bad format. == Saving '/etc/asterisk/voicemail.conf': Saved == Parsing '/etc/asterisk/users.conf': Found == Saving '/etc/asterisk/users.conf': Saved -- SIP/225-00719470 Playing 'vm-passchanged' (language 'en') -- SIP/225-00719470 Playing 'vm-options' (language 'en') Now, when I manually change the pin in the voicemail.conf file there is no problem. I tried looking on internet for any information, but I found nothing useful. Does anybody have any insight on why I can't change my voicemail pin via the Sip phone? Thanks in advance. Here is my voicemail.conf file: maxsilence = 10 silencethreshold = 128 maxlogins = 3 emaildateformat = %A, %B %d, %Y at %r tz = central dialout = outbound sendvoicemail = yes callback = outbound review = yes nextaftercmd = yes [zonemessages] eastern = America/New_York|'vm-received' Q 'digits/at' IMp central = America/Chicago|'vm-received' Q 'digits/at' IMp central24 = America/Chicago|'vm-received' q 'digits/at' H N 'hours' military = Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' european = Europe/Copenhagen|'vm-received' a d b 'digits/at' HM [default] 200 = 201 = 1234 225 = 1234 _ Find a local pizza place, movie theater, and more .then map the best route! http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theatercp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=950607encType=1FORM=MGAC01 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Password Issue
Seysan, I tried changing the DTMF format to RFC2833, but it did not help. Any other suggests? From: Seysan [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail Password Issue Date: Tue, 28 Aug 2007 15:26:47 -0600 Hello John, I think it is not the problem with your Asterisk, it is with your Phone (IP Phone or Softphone) Check the dtmf format on that. I think it is set to inbound, then change it to rfcxx. Then it should work fine. Regards, AFShin On 8/28/07, John Meksavan [EMAIL PROTECTED] wrote: Asterisk Users, I am running Asterisk-1.4.11 with Zaptel-1.4.4 on Debian Etch System 2.9.18-4-amd64. A TDM03B is installed on the Debian System. Every time, I try to change my voicemail pin via the Sip phone, the voicemail.conf does not get modify and I see this warning message on the Asterisk command line: [Aug 29 00:12:23] WARNING[19142]: app_voicemail.c:799 vm_change_password: variable has bad format. == Saving '/etc/asterisk/voicemail.conf': Saved == Parsing '/etc/asterisk/users.conf': Found == Saving '/etc/asterisk/users.conf': Saved -- SIP/225-00719470 Playing 'vm-passchanged' (language 'en') -- SIP/225-00719470 Playing 'vm-options' (language 'en') Now, when I manually change the pin in the voicemail.conf file there is no problem. I tried looking on internet for any information, but I found nothing useful. Does anybody have any insight on why I can't change my voicemail pin via the Sip phone? Thanks in advance. Here is my voicemail.conf file: maxsilence = 10 silencethreshold = 128 maxlogins = 3 emaildateformat = %A, %B %d, %Y at %r tz = central dialout = outbound sendvoicemail = yes callback = outbound review = yes nextaftercmd = yes [zonemessages] eastern = America/New_York|'vm-received' Q 'digits/at' IMp central = America/Chicago|'vm-received' Q 'digits/at' IMp central24 = America/Chicago|'vm-received' q 'digits/at' H N 'hours' military = Zulu|'vm-received' q 'digits/at' H N 'hours' 'phonetic/z_p' european = Europe/Copenhagen|'vm-received' a d b 'digits/at' HM [default] 200 = 201 = 1234 225 = 1234 _ Find a local pizza place, movie theater, and more .then map the best route! http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theatercp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=950607encType=1FORM=MGAC01 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ More photos, more messages, more storageget 2GB with Windows Live Hotmail. http://imagine-windowslive.com/hotmail/?locale=en-usocid=TXT_TAGHM_migration_HM_mini_2G_0507 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail Password Issue
Mojo, Thanks for helping me with this issue. You must have a NAME and EMAIL address after putting in the voicemail pin. I just migrate to Asterisk 1.4.x from 1.2.13, so I am still trying to get use to all the new stuff in the newer version. In Asterisk 1.2.13, it is not necessary to have a name and email address. Thanks again for your help in resolving this issue. Best Regards, John From: Mojo with Horan Company, LLC [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Voicemail Password Issue Date: Tue, 28 Aug 2007 14:10:19 -0800 While I can't say this won't work the way you have it, I CAN say it's not the way mine is set up and it's not a way I've SEEN it ever set up. Could it just be complaining that you've got nothing on the right side of the = for mailbox 200? Or could it be complaining that you don't have anything past the pin number on the other lines? Try: 201 = 1234,Name or 201 = 1234,Name,email I'm thinking it's my first suggestion, though. To test that, try adding another without a pin number: 199 = and see if you then get two of the variable has bad format error messages Moj John Meksavan wrote: Here is my voicemail.conf file: [default] 200 = 201 = 1234 225 = 1234 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ More photos, more messages, more storageget 2GB with Windows Live Hotmail. http://imagine-windowslive.com/hotmail/?locale=en-usocid=TXT_TAGHM_migration_HM_mini_2G_0507 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Experimenting- Sip dialing with Zap
Asterisk Users,
I have 3 FXO modules with the TDM400P Digium Card. I can dial into the
Asterisk rings my Sip phone, but dialing out with my SPA941 phone through
the zap channel is a problem. I keep getting this message on the Asterisk
CLI. What am I doing wrong? Thanks in advance.
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/200-006fa300, {Zap/g0/{EXTEN:1})
in new stack
[Aug 16 19:28:39] WARNING[14232]: channel.c:3209 ast_request: No channel
type registered for '{Zap'
[Aug 16 19:28:39] WARNING[14232]: app_dial.c:1106 dial_exec_full: Unable to
create channel of type '{Zap' (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/200-006fa300' status is 'CHANUNAVAIL'
I used sample I found the internet just to see if this is possible.
extensions.conf
[default]
exten = 101,1,Background(tt-monkeys)
exten = 1000,1,Dial(Sip/200,20,t,r)
exten = 1000,2,Voicemail(s1000)
exten = _XXX,1,Dial({Zap/g0/{EXTEN:1})
[telewest_pstn]
exten = s,1,Dial(Sip/200,25,t,r)
exten = s,2,Voicemail
exten = s,3,Hangup
zapata.conf
[channels]
context=telewest_pstn
switchtype=national
signalling=fxs_ks
rxwink=300
channel = 2-4
adsi=yes
usecallerid=yes
cidsignalling=bell
callerid=asreceived
hidecallerid=no
callwaiting=yes
usecallingpres=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
callreturn=yes
echocancel=yes
echocancelwhenbridged=yes
rxgain=00.0
txgain=00.0
group=0
callgroup=0
pickupgroup=0
echotraining=yes
immediate=yes
zaptel.conf
loadzone=us
defaultzone=us
fxsks=2-4
Best Regards,
John
_
Find a local pizza place, movie theater, and more
.then map the best route!
http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theatercp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=950607encType=1FORM=MGAC01
___
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Re: [asterisk-users] Experimenting- Sip dialing with Zap
After sending the email out, I went back to change the line in
extensions.conf from
Dial({Zap/g0/{EXTEN:1})
to
exten = _XXX,1,Dial(Zap/g0/{EXTEN})
I am using a phone simulator to test because I do not have the physical PSTN
line yet. The phone simulator only allow 3 digit dialing. Now, I get this
message on the Asterisk CLI
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/200-006fd1a0,
Zap/g0/{EXTEN})
in new stack
[Aug 16 20:22:34] WARNING[14292]: app_dial.c:1106 dial_exec_full: Unable to
create channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/200-006fd1a0' status is 'CHANUNAVAIL'
MY sip.conf file looks like this.
sip.conf
[general]
context=default ; Default context for incoming calls
bindport=5060 ; UDP Port to bind to (SIP standard port is
5060)
bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to
all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
tos=lowdelay;
lowdelay,throughput,reliability,mincost,none
;externip = 200.201.202.203 ; Address that we're going to put in
outbound SIP messages
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
;callerid=Delta Mobile Software 8478846728
; The Yoda VoIP wired phone drops audio when the codec is switched from
; ulaw to gsm after a ringback message is sent. As a workaround, only
; allow ulaw and alaw.
disallow=all
;allow=all
;mike's settings
allow=ulaw
allow=alaw
#include sip.generated
sip.generated
[200]
username=200
secret=yeengohh
type=friend
context=default
mailbox=200
callerid=200 GenNum 8478846750
host=dynamic
nat=no
canreinvite=no
From: James FitzGibbon [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Experimenting- Sip dialing with Zap
Date: Thu, 16 Aug 2007 12:49:16 -0400
On 8/16/07, John Meksavan [EMAIL PROTECTED] wrote:
CLI. What am I doing wrong? Thanks in advance.
The channel spec you need to use is:
Dial(Zap/g0/${EXTEN:1})
not
Dial({Zap/g0/{EXTEN:1})
Though bear in mind that the :1 is removing the first char of your
extension, so if you dial '123' on your Linksys, you'll dial '23' out your
analog line, which is unlikely to be what you want to do if said line is
connected to the PSTN. It's more typical to see something like
exten = _9NXXNXX,1,Dial(Zap/g0/${EXTEN:1})
which matches a 10 digit local number prefixed by nine, but removes the
leading 9 (using :1) because it's not needed (or wanted) by the telco.
--
j.
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Re: [asterisk-users] Experimenting- Sip dialing with Zap
David,
My TDM400P card is installed correctly or else, how could I dial into my
Asterisk box and make my sip phone ring? I do use the Asterisk List as the
last desperate option to solve any of my Asterisk problems. When all other
resources are exhausted and a different approach is in order to solve the
problem, then I turn to this list.
Thanks for taking the time to help me debug this issue.
Best Regards,
John
From: David Gomillion [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Experimenting- Sip dialing with Zap
Date: Thu, 16 Aug 2007 13:00:34 -0500
On 8/16/07, John Meksavan [EMAIL PROTECTED] wrote:
line yet. The phone simulator only allow 3 digit dialing. Now, I get
this
message on the Asterisk CLI
-- Executing [EMAIL PROTECTED]:1] Dial(SIP/200-006fd1a0,
Zap/g0/{EXTEN})
in new stack
[Aug 16 20:22:34] WARNING[14292]: app_dial.c:1106 dial_exec_full: Unable
to
create channel of type 'Zap' (cause 0 - Unknown)
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel 'SIP/200-006fd1a0' status is
'CHANUNAVAIL'
Just a guess here, but it looks like Asterisk is unable to create channel
of
type 'Zap', and that everyone is busy/congested at this time.
Now, figure out if you have valid Zap channels defined in both
zaptel.confand
zapata.conf. Make sure you have the right signalling, and the right
indications. Stupid question that I don't have to ask, but will anyway, you
do have the TDM400P actually installed, right?
With these basic questions, you may be better served reading a book about
Asterisk, trying what is in there, googling for answers to any questions
you
may have, and then asking the list after you have exhausted all other
resources. We're here to help, but I think that these steps may help give
you a better foundation. And we like it when people have at least tried to
figure out solutions.
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Re: [asterisk-users] FXO Modules and Sip Outbound
Erik,
In the sip.conf file, would I put my Asterisk Box's ip address in the
host field? What would I do with the registration field? Leave it alone?
Thanks in advance.
Best Regards,
John
From: Erik Anderson [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Subject: Re: [asterisk-users] FXO Modules and Sip Outbound
Date: Mon, 13 Aug 2007 16:36:08 -0500
On 8/13/07, John Meksavan [EMAIL PROTECTED] wrote:
Asterisk Users,
I have never done a dial plan for this scenario before. Is it
possible to
have Sip Phones make outbound calls through the PSTN? What would the
call
routing/dial plan would look like?
Yes - certainly possible. There's nothing different about the call
routing going from SIP-Zap as from SIP-SIP really. Assuming that
you already have your zaptel device(s) configured correctly, something
like this in your dialplan is all you'll need. This also assumes you
want to dial 9 to get an outside line.
[globals]
OUTBOUND-TRUNK=Zap/g0
[outbound]
exten = _9NXXNXX,1,Dial(${OUTBOUND-TRUNK}/${EXTEN:1})
-Erik
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[asterisk-users] FXO Modules and Sip Outbound
Asterisk Users, I have never done a dial plan for this scenario before. Is it possible to have Sip Phones make outbound calls through the PSTN? What would the call routing/dial plan would look like? -John _ Messenger Café open for fun 24/7. Hot games, cool activities served daily. Visit now. http://cafemessenger.com?ocid=TXT_TAGHM_AugHMtagline ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk Help
Asterisk Users, I am running Asterisk 1.2.13 on Debian Etch with McLeodUSA's T1 service. I have two Netgear switches on my T1 router, one for VOIP and another for data. I use a gigabit switch for all VOIP and a regular 10/100Mbps switch for all data. This morning I saw this message a few times on the Asterisk command line. The lagged cause garbled phone calls. Is my network to slow? Or is there something else going on? Aug 9 09:40:21 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1050ms / 1000ms) Aug 9 09:40:55 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:41:56 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:42:17 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:48:18 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:48:28 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:50:29 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:50:39 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:56:41 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1018ms / 1000ms) Aug 9 09:56:51 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 10:01:52 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 10:02:02 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Best Regards, John _ A new home for Mom, no cleanup required. All starts here. http://www.reallivemoms.com?ocid=TXT_TAGHMloc=us ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk Help
Gordon, Thanks for tip. Using this tool mtr makes it a whole lot easier to figure what is really going on. Thanks again. Best Regards, john From: Gordon Henderson [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk Help Date: Thu, 9 Aug 2007 17:09:05 +0100 (BST) On Thu, 9 Aug 2007, Paul wrote: I have the same debian and asterisk version combo running in more than one location. Some are T1 and some are in data centers. There have been times when I got such messages and some simple ping/traceroute testing showed obvious problems at my end or the provider end. Problems at the provider end were confirmed by testing from multiple locations. The 'mtr' command is handy here, although it can generate measurable traffic if you care to count every byte ;-) Just run mtr to the hostname of your upstream VoIP provider (netlogic?) and leave it running for a day or 2... If you don't have mtr, then: apt-get install mtr-tiny and you soon will have :) (mtr-tiny which is the text/curses version - the 'full' mtr is a GTK application, and running X applications on your asterisk box probably isn't what you want to do! Mtr *really* doesn't warrant a GUI IMO!!!) Gordon John Meksavan wrote: Asterisk Users, I am running Asterisk 1.2.13 on Debian Etch with McLeodUSA's T1 service. I have two Netgear switches on my T1 router, one for VOIP and another for data. I use a gigabit switch for all VOIP and a regular 10/100Mbps switch for all data. This morning I saw this message a few times on the Asterisk command line. The lagged cause garbled phone calls. Is my network to slow? Or is there something else going on? Aug 9 09:40:21 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1050ms / 1000ms) Aug 9 09:40:55 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:41:56 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:42:17 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:48:18 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:48:28 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:50:29 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 09:50:39 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 09:56:41 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1018ms / 1000ms) Aug 9 09:56:51 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Aug 9 10:01:52 NOTICE[6395]: chan_sip.c:9929 handle_response_peerpoke: Peer 'netlogic' is now TOO LAGGED! (1017ms / 1000ms) Aug 9 10:02:02 NOTICE[6395]: chan_sip.c:9923 handle_response_peerpoke: Peer 'netlogic' is now REACHABLE! (17ms / 1000ms) Best Regards, John _ A new home for Mom, no cleanup required. All starts here. http://www.reallivemoms.com?ocid=TXT_TAGHMloc=us ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ A new home for Mom, no cleanup required. All starts here. http://www.reallivemoms.com?ocid=TXT_TAGHMloc=us ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Failover Configuration
Asterisk Users, The more I work with the Asterisk 1.2.13 on the Debian Etch, the more realize there is no real reliable SIP provider. Having two Sip Providers is smartest thing to do, one being your main provider, while the other being the failover/safety. Ideally, I would like it to failover automatically when there is congestion/busy or sip channel dead lock. If I have to press 9 or some random digit just to make a call out on the 2nd Sip provider, it would be fine also. What would be the best way of to doing this? Trying dialing 9 first to use the netlogic trunk did not work, so I commented it out and tried doing the following way. This does not work. Any Suggestions? [bv-outbound] exten = _1NXXNXX,1,Macro(dialout|SIP/[EMAIL PROTECTED],30) ;exten = _91NXXNXX.,1,Macro(dialout|SIP/[EMAIL PROTECTED],30) exten = _1NXXNXX,2,Congestion exten = _1NXXNXX,3,Macro(dialout|SIP/[EMAIL PROTECTED],30) Best Regards, John _ Now you can see trouble before he arrives http://newlivehotmail.com/?ocid=TXT_TAGHM_migration_HM_viral_protection_0507 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VOIP Provider- Callcentric
Asterisk Users, I am looking for Sip Providers for my Asterisk 1.2.13, running Debian Etch system with McLeodUSA's T1 service. Has anybody ever used Callcentric for their Sip Provider? Any service issues with Callcentric? Best Regards, John _ Messenger Café open for fun 24/7. Hot games, cool activities served daily. Visit now. http://cafemessenger.com?ocid=TXT_TAGHM_AugHMtagline ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] VoicePulse Connect
Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] VoicePulse Connect
Wes, What kind of service outages did you experienced? This would use for my office and I cannot afford for any dropped calls or poor audio quality, when talking to customers. -John From: Wes Baehr [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: 'Asterisk Users Mailing List - Non-Commercial Discussion'asterisk-users@lists.digium.com Subject: Re: [asterisk-users] VoicePulse Connect Date: Wed, 8 Aug 2007 12:55:29 -0400 John, Voicepulse Connect has been great to me. I've been using it for over a year now, and do not have any major complaints, except that there are no printable receipts for credit card transactions. SIP is also the preferable protocol, as IAX seems to have some issues. Customer service is usually pretty good, and there have been very few (although a couple) problems with service outages. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of John Meksavan Sent: Wednesday, August 08, 2007 12:30 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] VoicePulse Connect Asterisk Users, Has anybody use Voicepulse Connect for Asterisk? I am trying to cover all my bases because in the past, I got burned with poor quality of service, along with failed DTMF tones with 3 different SIP Providers (Vitelity, Broadvoice, and Teliax). I am running Asterisk 1.2.13 on the Debian Etch system, using the SIP protocol. Any insights would be great. Thanks. -John _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Find a local pizza place, movie theater, and more .then map the best route! http://maps.live.com/default.aspx?v=2ss=yp.bars~yp.pizza~yp.movie%20theatercp=42.358996~-71.056691style=rlvl=13tilt=-90dir=0alt=-1000scene=950607encType=1FORM=MGAC01 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Teliax Quality of Service
Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? It seems like when I am ready to go live with my Asterisk PBX System, I run into quality of service issues with the SIP provider. Who should I go with that would guarantee me quality service just like an analog line? _ See what youre getting into before you go there http://newlivehotmail.com/?ocid=TXT_TAGHM_migration_HM_viral_preview_0507 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Teliax Quality of Service
Asterisk Users, In my setup, I have a T1 service with McleodUSA and I am using the SIP protocol. I am considering switching back to analog lines because quality of service outweighs the cost savings at my work. Any good SIP providers out there? From: Baji Panchumarti [EMAIL PROTECTED] Reply-To: [EMAIL PROTECTED],Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Teliax Quality of Service Date: Thu, 2 Aug 2007 16:40:27 -0400 On 8/2/07, John Meksavan wrote: Asterisk Users, I recently ran into some problems with the quality of service with Teliax. This occurred on August 1, 2007 with a dropped outbound call, audio quality isse on the callee side- not hearing me well on callee side, and sending DTMF tones (configured for RFC2833). Am I the only Teliax customer having this problem? ditto here this week, random breaks in audio, garbled voice etc. My softphones dialing in from outside had no audio issues. Others on teliax forums suggested I switch to SIP since iax2 is aggressively evolving and teliax equipment is experiencing some incompatibilities with recent * iax releases. I changed codecs from gsm to ulaw, voice quality improved but same random breaks. It seems like when I am ready to go live with my Asterisk PBX System, I run into quality of service issues with the SIP provider. Consider having some fall back options from alternate providers since it doesn't cost a whole lot to keep an active account. Who should I go with that would guarantee me quality service just like an analog line? I have heard that there is no such thing unless your provider you have a dedicated, or at least highly reliable, circuit between the two of you : http://en.wikipedia.org/wiki/User_Datagram_Protocol UDP does not guarantee reliability or ordering in the way that TCP does. Datagrams may arrive out of order, appear duplicated, or go missing without notice. Avoiding the overhead of checking whether every packet actually arrived makes UDP faster and more efficient, at least for applications that do not need guaranteed delivery. Time-sensitive applications often use UDP because dropped packets are preferable to delayed packets... One of the reasons Time Warner, Armstrong, Cox and other cable broadband guys are able to offer fairly reliable voip service is that they control the pipes between their VoIP proxies and their end users. It is also the reason vonage, teliax and other 3rd party vendors have more issues. I used broadvox a few years ago, if the callee answered before the caller had heard a ring the line went dead :-) -baji. -- ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Tease your brain--play Clink! Win cool prizes! http://club.live.com/clink.aspx?icid=clink_hotmailtextlink2 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk DTMF Tones
Asterisk Users, I am running Asterisk 1.2.13 on Debian Linux 2.6.18-4-amd64 and having problems with DTMF Tones. I have sip service from Teliax and configure to use rfc2833 for dtmfmode. The problem occurs, when I am using Linksys PAP2T phone adapter with a regular analog phone. Is this an issue with Asterisk? Or the Linksys PAP2Any insights would be greatly appreciated. Best Regards, John _ http://liveearth.msn.com ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk DTMF Tones
Alex, The DTMF tones are being sent twice. On SIP Peer side, I set the DTMFMODE=RFC2833 and the PAP2T you can choose from INBAND, AVT, INFO, and AUTO, so I chose Auto. Should change on Peer Side and the PAP2T side to use INBAND? From: Alex Balashov [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk DTMF Tones Date: Wed, 1 Aug 2007 15:51:48 -0400 (EDT) John, On Wed, 1 Aug 2007, John Meksavan wrote: I am running Asterisk 1.2.13 on Debian Linux 2.6.18-4-amd64 and having problems with DTMF Tones. I have sip service from Teliax and configure to use rfc2833 for dtmfmode. The problem occurs, when I am using Linksys PAP2T phone adapter with a regular analog phone. Is this an issue with Asterisk? Or the Linksys PAP2Any insights would be greatly appreciated. What exactly are the issues? And, did you set the dtmfmode to RFC2833 for the SIP peer that corresponds to the Linksys PAP2T as well? Does it support out-of-band DTMF? Thanks, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ http://imagine-windowslive.com/hotmail/?locale=en-usocid=TXT_TAGHM_migration_HM_mini_2G_0507 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk DTMF Tones
The DTMF tones are being sent twice. On SIP Peer side, I set the DTMFMODE=RFC2833 and the PAP2T you can choose from INBAND, AVT, INFO, and AUTO, so I chose Auto. Should change on Peer Side and the PAP2T side to use INBAND? From: Alex Balashov [EMAIL PROTECTED] Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk DTMF Tones Date: Wed, 1 Aug 2007 15:51:48 -0400 (EDT) John, On Wed, 1 Aug 2007, John Meksavan wrote: I am running Asterisk 1.2.13 on Debian Linux 2.6.18-4-amd64 and having problems with DTMF Tones. I have sip service from Teliax and configure to use rfc2833 for dtmfmode. The problem occurs, when I am using Linksys PAP2T phone adapter with a regular analog phone. Is this an issue with Asterisk? Or the Linksys PAP2Any insights would be greatly appreciated. What exactly are the issues? And, did you set the dtmfmode to RFC2833 for the SIP peer that corresponds to the Linksys PAP2T as well? Does it support out-of-band DTMF? Thanks, -- Alex Balashov Evariste Systems Web: http://www.evaristesys.com/ Tel: +1-678-954-0670 Direct : +1-678-954-0671 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _ Need a brain boost? Recharge with a stimulating game. Play now! http://club.live.com/home.aspx?icid=club_hotmailtextlink1 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Sip Providers
Asterisk Users, I have Asterisk PBX System running at my work. The system is working great. Currently, I have Broadvoice as my sip provider and I am not completely satisfy with their service. Broadvoice only allows 2 simultaneous calls, which hinders my company's communications ability. I am looking for a sip provider that would work with Asterisk and allow at least 6 simultaneous calls, locally and internationally. Of course the voice quality, pricing, number portability are the main determining factors. I will have a T1 connection at the office, so bandwidth would not be an issue. Any thoughts on this matter would be greatly appreciated. Thanks. Best Regards, John _ Need a brain boost? Recharge with a stimulating game. Play now! http://club.live.com/home.aspx?icid=club_hotmailtextlink1 ___ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
