similar to: question about connecting with Mobile Base Station

Hi, I have a Linksys SPA 3102 using as ATA. The call routing is : Phone -> PSTN -> SPA 3102 -> SIP Proxy -> Asterisk The problem I am having is that when the phone hangs up, SPA 3102 can't detect it and relay the CANCEL message. Is this problem with my SPA 3102 config or it just works like that by default? Thanks in advance for your help. Regards, Mark -------------- next

Hi, I have two asterisks that are connected to each other via a back-to-back E1 link using a pair of sangoma cards. With the following scenario: SIP-PHONE <-> Asterisk <-> E1 <-> Asterisk <-> SIP-PHONE, the sound quality degrades significantly. I can't understand why as the amound of packet lost should be very minimum. Does anyone know why? Does it have anything

Hi I am working on deploying voip for my company and would like to seek some advice on the number of E1 lines we need to rent. Our telco told us that there can be at most 30 concurrent channels on an E1 line. Typically, what is the maximum number of DIDs that we can allocate to that E1 line before users get frequent "all lines are busy"? We are running a support center with mostly

Hi all, I am using Digium PCI board to receive PSTN call through regular phone line. It is no problem for me to receive calls, but I am not able to obtain the Caller ID if the calls are from the phone line. exten => s,1,Answer() exten => s, n, Verbose(1|incoming number is ${CHANNEL} calling to ${EXTEN} routing to ${phonenum} ) exten => s,n, Verbose(1|callid is ${CALLID(num)}) exten

Dear friends, I am having problem with running a sample php and I can't figure out why. I can run the sample.php using CLI but when I run it inside the dialplan it does not work. Can someone please suggest the config problem that I may have made? dommy:/var/lib/asterisk/agi-bin# php sample.php #!/usr/bin/php5 -q VERBOSE "Here we go!" 2 VERBOSE "Call from - Calling

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

Is anyone aware of a GSM femtocell that will trunk back to a VoIP softswitch such as Asterisk? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100802/4f58fea7/attachment.htm

I want to know about asterisk and openBTS If anybody made any test and experience... Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100819/cbd08740/attachment.htm

Friends, After listening to Mark Summer's keynote at Astricon (hopefully soon on the Astricon web site) I think we should come back to the discussion he started. Mark talked about using Open Source in general and Asterisk in particular in third world projects as well as in emergencies in other countries. He and Inveneo help groups of people to get a better understanding of how to build

I'm trying to use OpenBTS with Asterisk. I have two phones that are connecting to OpenBTS correctly, but on the Asterisk side the phones can't call each other. I followed this guide: http://gnuradio.org/redmine/projects/gnuradio/wiki/OpenBTSSettingUpAsterisk I set up two phones in sip.conf and extensions.conf. In my SIP output I see this: WARNING[1689]: app_dial.c:2041 dial_exec_full:

Hi, I am looking for a very low cost way of receiving and sending T38 fax reliably. Is there any possible solution using Asterisk as the PSTN SIP gateay and Digium E1/T1 card? Is there other open source package that can help to accomplish this purpose? Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I met a problem in asterisk, please see message in the following, the detail debug log is in the attached file. can someone help to point out where to correctly configure asterisk, thanks a lot ! BR/Scott -------> -- Executing [8690 at phones:1] Dial("SIP/IMSI466990004244439-00000014", "SIP/IMSI466974104638690") in new stack Really destroying SIP dialog '

Hi all, This is not a repeated post as I am just adding more information for my previous post. Asterisk version 1.4.18 TDM card: Digium TDM411B Zaptel version 1.4.9.2 Line: PSTN line I tried to obtain the dialed number using $DNID and $CDR(DST) . All of these variable returns 's' I also tried exten => _3345335,n,Noop(this is ok) where 3345335 is my number but it does not go there.

Hi all, I am using Digium PCI board to receive PSTN call through regular phone line. It is no problem for me to receive calls, but I am not able to capture the destination number through the ZAP channel exten => s, n, Verbose(1|destination to ${EXTEN} ) ${EXTEN} returns 's' instead of the actual destination number. Since I have multiple phone numbers, I want to be able to route

Does anyone have a copy of spc.exe they could send me? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061117/ac0b6a44/attachment.htm

Anyone on this list have extensive experience with BTS? http://deancollinsblog.blogspot.com/2009/03/open-bts.html Please email me, particularly if you have experience in deploying over multiple cells covering large geographical areas (200k's sq). Regards, Dean Collins Cognation Inc dean at cognation.net <mailto:dean at cognation.net> +1-212-203-4357 New York

I wish to use AsteriskNOW (the Digium repository + CentOS) with imap voicemail storage and Asterisk 1.4. After having installed AsteriskNOW with Asterisk 1.4 and Asterisk GUI I run the yum package manager and replace voicemail with imap voicemail and attempt to start Asterisk, however the voicemail module is not loaded: [Mar 23 23:30:03] WARNING[12592]: loader.c:382 load_dynamic_module: Error

Does anyone know what version of Asterisk Switchvox uses, and if it is modified in any way? FWIW, I am dealing with a provider that claims compatibility with Switchvox but not Asterisk for their SIP trunking service.

Hi, I have read about SS7 recently and learnt that it is a signalling protocol used in PSTN for call management, setup, etc. The thing that I don't understand is how SS7 plays a role in VOIP. When I make calls between landline and Asterisk via PSTN, I don't need to do anything with SS7. Is it because the SS7 signalling is already done by Asterisk already? From the prespective of

Hi list, I have a problem with installing Asterisk (under https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages): sudo apt-get install asterisk-1.8 Reading package lists... Done Building dependency tree Reading state information... Done Note, selecting 'asterisk' instead of 'asterisk-1.8' Some packages could not be installed. This may mean that you have requested an