similar to: [asterisk-dev] Asterisk + OpenIMSCore

... Activating "sip debug" shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named

tnet.itand SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo < gincantalupo at

Hi, The problem solved After installing new zaptel drivers, we ran the "genzaptel" command to generate /etc/zaptel.conf file,checked with "zttool" command and the card status was "Yellow alarm/Blue alarm/Recovering" and the card LED was blinking red and green. The problem was with the generated zaptel configuration., but not with the pin

Or even: <a class="moz-txt-link-freetext" href="http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&mid=4946">http://www.blackbox.com/Catalog/Detail.aspx?cid=425,1423,1424&amp;mid=4946</a> (same thing from the UK site:) <a class="moz-txt-link-freetext"

ast_waitfordigit that accepts milliseconds as input. Douglas Garstang wrote: > Admittedly I have not used the ExternalIVR app. Is it any good? > > I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, > it can do it, but boy it is UGLY. There's also the fact that you can't > call Backgound() in a macro, which forces you to use Read() which >

<br> Or even:<br> <br> &nbsp;<a href=3D"http://www.blackbox.com/Catalog/Detail.aspx?cid=3D425,1423= ,1424&amp;mid=3D4946" target=3D"_blank">http://www.blackbox.com/Catalog/Det= ail.aspx?cid=3D425,1423,1424&amp;mid=3D4946</a><br> <br> (same thing from the UK site:)<br> <br> <br> <a

an external program, which at this stage, is not customizable ... I don't know if alternatives (LiMO, Android, ...) would be more open to this customization but for Symbian, not only Nokia SIP client wouldn't let you autoanswer to SIP calls, but any other SIP client complying to Symbian design wouldn't support autoanswer. PS: Please, note that I'm far from being an expert in GSM

mailboxes). Are you certain that removing either 612 or 610 mailbox would keep Asterisk from complaining ? > > However, the MWI does not indicate voice mails for 610 and I keep seeing > this error message: > > ERROR[2549]: app_voicemail.c:1630 messagecount: Couldn't find mailbox > 610 in context a10 > > However, mailbox 610 is clearly defined in voicemail.conf: >

2008/9/29 Alex Balashov <abalashov at evaristesys.com> > Try this: > > exten => _XXXX,1,Set(THISTECH=${CUT(CHANNEL,/,1)}) > exten => _XXXX,n,NoOp(Technology is ${THISTECH}) > exten => _XXXX,n,Set(THISCHANNEL=${CUT(CHANNEL,/,2)}) > exten => _XXXX,n,NoOp(Channel is ${THISCHANNEL}) Hi, I don't have any spare zaptel enabled system I could try this on, but I

I get congestion (same error) with exten => _NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r) not dialing 1 exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r) dialing 1 exten => _91NXXNXXXXXX,1,Dial(SIP/${EXTEN:1}@xxx.xxx.xxx,30,r) dialing 9 All the same == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [9544790554 at To_Airspring:1]

That sounds like the ideal app for me too. Fring requires we register with Fring and give them user id/password pair. In our case it did not work until we put a public IP on our Asterisk. I just bought WeePhone and I'll give it a try on the iPhone. Cheers, Enrique -----Mensaje original----- De: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] En

I'm using it rather successfully. Not perfect, but it works. It is limited to WiFi connectivity... at least here in Spain I cant get either client to work over 3G. I'm using Fring and Truphone. Although I have only configured a SIP to my Asterisk with Fring. Skype works fine. We tested with several Nokia 5800 (EM) using Fring. Call quality is worse. At best, we have a 1+ second delay.

Astricon is only 54 days away! If you're not booked, please take a moment to register for the conference, get your hotel room, and get your plane tickets before things fill up and/or get expensive. This is a great opportunity to meet other developers, users, and members of the Asterisk ecosystem, and I encourage everyone to attend. While there are great things to be said about the

<br> Or even:<br> <br> &nbsp;<a href=3D"http://www.blackbox.com/Catalog/Detail.aspx?cid=3D425,142= 3,1424&amp;mid=3D4946" target=3D"_blank">http://www.blackbox.com/Catalog/De= tail.aspx?cid=3D425,1423,1424&amp;mid=3D4946</a><br> <br> (same thing from the UK site:)<br> <br> &nbsp;<a

1. http://bugs.digium.com/view.php?id=12362 2. http://bugs.digium.com/view.php?id=12925 3. http://bugs.digium.com/view.php?id=12921 Also how do you go about changing details for device in DB and not using "sip realtime prune PEER" + 'sip reload'? Without that your changes to devices are not active. Good luck! Regards, Mindaugas Kezys http://www.kolmisoft.com >

Hello Nhadie, I had a very similar situation. My solution, even tough might not look very wise, solved my problem the way I needed. I repeated the GotoIftime command in the next line in my extensions.conf . Like this: GotoIfTime(22:00-23:59|*|30|jul?test,s,1) GotoIfTime(00:00-02:00|*|31|jul?test,s,1) Rgs, Marco Cordeiro -----Mensagem original----- De: asterisk-users-bounces at

Hey Guys, I am trying stream live music via icecast streaming server into a conference room, this will allow persons joining the conference to hear the music. I have been googling and i have come across a few tutorials, that give instructions as to how to get it done. But they all mention the use of a ices application module. It appears that asterisk 1.4 is not shipped with app_ices.0 by

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the CNAM info in the Q.931 call setup message. I've tried all permutations of switchtype (dms100 & national) and facilityenable that I can think of, but I still don't see CNAM coming out the other side. Telco confirms that "Name Out" is enabled on our PRI. Any pointers on what I'm missing, and/or how to debug further? zapata.conf: --- [channels] context=3Ddefault

MD -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Michiel van Baak Sent: September 12, 2008 2:08 PM To: Asterisk Users List Subject: Re: [asterisk-users] Setup speed dials on Cisco 7921 On 13:15, Fri 12 Sep 08, OCG Technical Support wrote: > I've added lines like this: > > > >