search for: millican

...hing works as it should. moved 1.2.0 configs to 1.2.6 box and again no audio between sip channels. *CLI> sip debug SIP Debugging enabled *CLI> <-- SIP read from 192.168.1.200:5060: INVITE sip:699@192.168.1.10 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.200:5060;branch=z9hG4bK-75990941 From: John Millican <sip:677@192.168.1.10>;tag=f250a44bc61492b1o0 To: <sip:699@192.168.1.10> Call-ID: 23a92fd3-a3463cc9@192.168.1.200 CSeq: 101 INVITE Max-Forwards: 70 Contact: John Millican <sip:677@192.168.1.200:5060> Expires: 240 User-Agent: Linksys/PAP2-2.0.12(LS) Content-Length: 235 Allow: ACK,...

I am trying to setup one time caller id block on my system(activated when an incoming call matches *811XXXXXXXXXX), and I have had little to no luck. Could you take a look at my context/macro definition and help me figure out what I am missing? Here is my context for my dialplan: include=default plancomment=user-default

...ript /var/lib/asterisk/agi-bin/callid.c -- Playing 'abandon-all-hope' (language 'en') == Spawn extension (default, 657, 4) exited non-zero on 'Zap/1-1' agi_request: callid.c -- Hungup 'Zap/1-1' What am I doing Wrong? Thank you very much for any help John Millican (a newbie obviously) --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.740 / Virus Database: 494 - Release Date: 8/16/2004

Hello all, I have seen some chatter again about DTMF. I see most of the talk about DTMF around not being able to get an external IVR to recognize digits, not a big issue for me at this time but sill interesting. My issue though, is with talk off on a zap channel. It seems to be getting worse or maybe my patience is thinning. All my calls go out and come in pstn through an FXO as I do not

Hello, I am just curious how many people are hooked up with BroadVoice and have recently been experiencing a lot of dificulty. Joel

Hello all, OS OpenSuSE 10.3 * ver 1.4.26.2 zaptel ver. 1.12 Digium TE122 I have a request for remote users to be able to dial through the system so that the sales managers can barge/chanspy on the sales force. I have the DISA part working with authentication(rather straight forward) but what I can not figure out is how to enable the supervisors to be able to barge on these calls. Is there a

Hello all, Okay when you are done laughing at the simplicity of this question could someone show me please what I have wrong in the following statement? GoToIf($[${numdial} != [1-9] ]?15:3); What this is supposed to do is if numdial is not a single digit from 1 to 9 inclusive goto 15, if it is a singledigit from 1 to 9 inclusive goto 3. Should be pretty simple but not working for me, always

Hello all, I am looking into building a calendar server (due to business requierments I can not use public hosted calender like Google), and am looking for suggestions based on experience with different calendar applications/servers available for Linux that you have integrated with Asterisk. If you can give a quick, simple list of what worked and what didn't I would be very grateful.

Hello All, I am looking at doing some video conferencing with SIP. I was hoping to get some early pointers from any one that is currently doing this. I have been all over goggle and voip-info and there is a ton of anecdotal information but, I was hoping for more specifics of what people are actually using that works and even some of what hasn't worked so that I can stay away. What I am

I receieved my first PAP2-NA yesterday from our distributor(Tech Data). It installed pretty easily and has worked great so I went to order some more of these units today. When I logged into Tech Data this morning, the PAP2-NA was now marked as discontinued and no longer available and only the PAP2 version was available which is the Vonage branded version. :( I saw someone on the list say that

As I did two years ago, "I'm posting a new thread with the "Top Posting" subject" rather than hijacking the "Paging for Praying" thread. Two questions: 1. Steve K: What do you mean by "/coat"? 2. How do we change rule #5? --Don Don Kelly PCF Corp People Come First 651 842-1000 651 842-1001 fax -------------- next part

Hi, I'm using the Read command the read a DTMF tone. In this read command I play a voice-file. But now when I press one off they keys of my telephone the voice-file will stop playing a the program go the next priority. Is it possible to play the voice-file until the right DTMF tone is pressed? (say for instance the Zero). Kind regards Arjan Kroon Mobillion B.V.

Hello All, I have googled and wikied but must not be searching correctly. Assuming the TE110P has same ability as old T100P to use some voice and some data channels, lets say I have a TE110P set to accept voice on 10 channels and pass the other 14 channels as data. Under this scenerio i am "guessing" that * should still be able to accept VoIP calls on the data channels and still

Dear Group, I have just purchased two Linksys SPA941 and flashed these to the latest firmware. Everything works well except for the Hold button? Has anyone else experienced the same issue? What was the solution? Kind Regards Shad Mortazavi

Hello All, I have Asterisk 1.4.5 running on a SuSE 10.3 x86_64 2.6.18.2-34 I upgraded from 1.4.2 to 1.4.5 on sunday the 24 of june and since have been getting: Internal RTCP NTP clock skew detected: lsr=1402479300, now=1402675136, dlsr=196500 (2:998ms), diff=664 I see an entry in Mantis that Russell fixed code so that this will not show when it shouldn't. Would i be correct in

Hello All, I have a requirement to setup a predictive dialer for a customers call center. I am asking for pros and cons of the different dialers available for Asterisk. If you are going to send marketing material send it to my e-mail directly please and not to the list. I was hoping to get the opinions of any one using any of these dialers and what they liked and didn't like, ease of

Hello All, I have a need for a wireless solution and have been looking at the Aastra 57i CT phone that have the wireless handset with them. Aastra says they will cover "up to 300,000 square feet". I am finding this hard to accept. I was also wondering about the "secure WDCT cordless technology" Could this be a form of DECT? Any one using these that can shed some lite?

Hello all, I know this has been covered on list but can not find the answer I need, lots of references to no authority found, but none with an answer. I have two * servers, one behind firewall with nat the other on a dmz with nat. Both servers register with each other successfully. home is today's CVS-HEAD away is Asterisk 1.0.7 on away: Registered to '165.xxx.xxx.xxx', who sees

Hi all! I am looking for some software that would work as a proxy between a SIP device (SIP phones and ATA boxes) and the Asterisk system running IAX. The reason is that I can only get IAX trow the firewalls, and can't change the settings. One solution I am using are getting several Asterisk system communicate trow IAX bout in this case would I rater have every persons computer act as a proxy

Hello all, Ok it is Sunday afternoon and I am going crazy. I have been running in circles so long that I can't think straight. As an example, I sent this message to the wrong address the first try, AAARRRRGGGGGGHHHHH. I have Asterisk 1.6.0.6 and DAHDI Tools Version - 2.1.0.2, DAHDI Version: 2.1.0.4, OpenSuSE 10.3 x86_64, tdm422 at the end of installing dahdi-linux and dahdi-tools I get: