similar to: Dial Plan Problems (also IAX)

Displaying 16 results from an estimated 16 matches similar to: "Dial Plan Problems (also IAX)"

2003 Mar 06
2
Dial Problem
I have a simple problem with sialing a SIP device. I'm SURE it's a syntax problem, but I dunno what it might be. Here are the debug messages: == Accepting call on 'Zap/1-1' ("PENSACOLA, FL" <8503846785>) -- Executing Goto("Zap/1-1", "2111|1") in new stack -- Goto (default,2111,1) -- Executing Dial("Zap/1-1",
2003 Jun 13
2
Asterisk asterisk => statement
As I understand it (and my understanding is obviously incorrect) the switch => statement sells the Asterisk box to resolve (aka lookup) extensions by querying the remote Asterisk server defined in the switch => statement. The switch => statement is used to centralize dialplans. I've not used the switch => statement yet, I'm just trying to understand the ramifications of using
2003 Mar 06
1
Cisco SIP Weirdness (1750, not ATA)
I have the following in extentions.conf: exten => 2111,1,Dial(SIP/2111 at gw1.langley) exten => 2111,2,Voicemail(u2111) exten => 2111,3,Hangup exten => 2111,100,Voicemail(b2111) exten => 2111,101,Hangup I have the following in sip.conf: ; Cisco 1750 [gw1.langley] type=friend host=172.16.17.1 context=default canreinvite=no Like the ATA, lots of stuff doesn't work on the 1750
2003 Apr 10
1
Problems compileing latest CVS
I'm getting the following message when I try to compile asterisk DEBUG[1024]: File config.c, Line 653 (__ast_load): Parsing /etc/asterisk/modules.conf WARNING[1024]: File loader.c, Line 212 (ast_load_resource): /usr/lib/asterisk/modules/chan_sip.so: undefined symbol: ast_pickup_ext WARNING[1024]: File loader.c, Line 319 (load_modules): Loading module chan_sip.so failed! I totally deleted my
2003 Oct 03
2
Transfer from IAX call
I am using IAX to send a call to my cell phone. I want to be able to hit # and transfer it back into the office. I have added tTr to the dial command and hitting # prompts me for the transfer, but after I start dialing 103, it stops at 1 and tries to transfer it within nufone instead of my dialplan. This is the debug output: -- Called me@NuFone/1515480XXXX -- Call accepted by
2011 Feb 24
1
Registration failed though configured.
Hi list, Currently, one of my phones registers fine, and the other does not, though for me they have the same config... Can somebody help debug/understand why? The logs in asterisk say: [Feb 24 13:48:09] NOTICE[20626]: chan_sip.c:15642 handle_request_register: Registration from 'IMSI208300618462231 <sip:IMSI20830061xxxx at 127.0.0.1>' failed for '127.0.0.1' - No matching
2003 Apr 13
1
Playback application
What file formats does the Playback() application support? "show application playback" does not say. I'm trying to playback a .wav file and it's not working. --Eric -- BTEL Consulting 504-595-3916x2111 (Experimental) 850-484-4535 (Office) 877-552-0838 (Cell)
2003 Oct 26
1
NuFone International Calls
Does anybody know how to do an international call using NuFone. I realise this isn't really the place to ask, but NuFone appears to be closed for the weekend and would like to have a try at this before tomorrow. I assumed it would be '011' for an international line followed by country code but that doesn't seem to work. I am getting: -- Executing
2008 Feb 25
4
TDM400P dialout problem
Using asterisk 1.4.18 and zaptel 1.4.9 on x86_64, I am having trouble dialing out to the pstn. The call is initiated at Zap/1-1 and should exit via Zap/3. I get the following: -- Starting simple switch on 'Zap/1-1' -- Executing [2111 at internal:1] Dial("Zap/1-1", "Zap/3/8801234") in new stack [Feb 25 02:36:59] DEBUG[7194]: chan_zap.c:1954 zt_call: Dialing
2011 Jan 10
0
No subject
do not know why. Anybody has a clue what could be wrong ? Is this a bug ? [I rebooted asterisk, and now it works.] Regards Axelle. Logs of failed registration: > sip show users Username Secret Accountcode Def.Context ACL NAT IMSI208011234567890 sip-local No RFC3581 IMSI208302141472352 sip-external No
2003 Jul 09
1
PBX / Asterisk integration
Hi all, I regret that I don't know much about telephony as I'm a networking bod, but here goes... We are thinking about implementing a VoIP service so that staff and students can make VoIP calls from home or using our wireless LAN on campus. Clearly, we would like it to integrate with our PBX so VoIP users can talk to the PSTN as well. We don't actually control the University's
2006 May 26
1
End of migration: adding support for some an alog phones
Nuthin beats an Atlas: http://www.adtran.com/adtranpx/Doc/0/TUA2HMOPDK3KN6S9LM1FH91169/61200305L2-8 A.pdf Telephony Swiss army knife. You can make it do anything. Be prepared to crap your pants when you see the price, though. -----Original Message----- From: Time Bandit [mailto:timebandit001@gmail.com] Sent: Friday, May 26, 2006 9:23 AM To: Asterisk Users Mailing List - Non-Commercial
2009 Jan 19
1
Suggestions on how to create a hunt or hunt like (rollover, multi-line) group or where to get one?
I have about 5 incoming USA SIP lines, but my provider does not have any sort of roll-over or huntgroup feature. Does anybody have an idea on how I can create a general number that will ring to the next available, non-busy SIP line that I have? Is there a provider out there that would do this? Any suggestions would be greatly welcome. Thank you. -------------- next part --------------
2010 Mar 29
3
Slightly more advanced dialling..
Hello, I'm wondering if it is possible to ring X number of extensions simultaneously, and each answered call can be handled with some code. I can do a huntgroup-esque way of dialling, but I want all the dialled numbers to be picked up. I hope this makes sense.. If not please say.. Many thanks! Andy -------------- next part -------------- An HTML attachment was scrubbed... URL:
2009 Feb 12
4
Asterisk Queue and URL Calling
Dear All I want to integrate sugarcrm and asterisk , so when customer call the call center the agent or extension which answers the call , before pickup the phone and talk to customer , view his/her information if it is available. I do this as described below : 1-Setup login username for sugarcrm for each extension 2-Extension Users will login to the sugarcrm. 3-Develop php script to handle new
2005 Feb 08
11
More complicated huntgroups / delayed ringing
Stefan Gofferje wrote: > Hi Folks, > > on my home asterisk, I have a "huntgroup" for incoming calls on the > private line which first let ring my phones in my office and living > room, after a while then office, living room and bedroom. > I do this by simply putting two dial statements in sequence: > > > [private_huntgroup_day] > exten =>