similar to: pipemedia

Has anyone out there used the PipeMedia/PipeCall PSTN gateway? Anything good/bad to say about it? I'm considering using them for a new customer. They seem to have good rates, good provisioning tools and (better still) give commission on usage to dealers. -- David Gurr Congruity Ltd. Fax: 0871 661 1756 Hemel Hempstead UK

Dear, I added new field to cdr table , named "service" and type varchar(20), but in extensions.conf with the following command, nothing to be saved. exten => _X.,1,Set(CDR(service)=OUT) does asterisk support this ability ? is any setting must be changed, before that ? best Mani

hello is any web based sip phone? for example: a user after logining in, view a configured sip phone, and ...... best MAni ____________________________________________________________________________________ Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains. http://farechase.yahoo.com/promo-generic-14795097

Dear, is any way to change , the size of voice packets? I want to increase the quality by decreasing the size of each packets, because of bandwidth failure. ? thanks in advance Mani -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081110/c1b2ed9d/attachment.htm

Hi I have installed, asterisk , with postgresql. it 's the view of extensions table: didex=# select * from extensions order by id desc limit 5; id | context | exten | priority | app | appdata | description

Dear, Our iax clients's ip and port in the database were removed automatically, after 30 secs. the iax info is saved in odbc and postgresql . asterisk=# select * from iax_buddies where username='9706015'; name | username | type | secret | md5secret | dbsecret | transfer | inkeys | outkeys | auth | accountcode | amaflags | callerid | context | defaultip | host | language

Dears my Internet Provider , prevents , sip connections, between sip client(sip phone) and sip server, (asterisk + ser) . both of client and server are mine. is there any solution for tunneling the sip packets? best Mani ____________________________________________________________________________________ Don't pick lemons. See all the new 2007 cars at Yahoo! Autos.

hi After compiling and installing pwlib and openh323 , the asterisk, give the folloing error. please tell me where the problem is ? Best Mani *CLI> -- Executing Dial("SIP/2.2.2.2-086f5ac0", "H323/652#150388590962@1.1.1.1|60") in new stack Mar 28 14:17:23 WARNING[11985]: channel.c:2576 ast_request: No translator path exists for channel type H323 (native 4) to 256 Mar 28

Hi I has installed, asterisk , with postgresql. it 's the view of extensions table: didex=# select * from extensions order by id desc limit 5; id | context | exten | priority | app | appdata | description

Hi Dear I want to use asterisk(1.2.7.1) as a router by caller id. I have only a DID number, I want to map this number to some ip-phones , base on received Caller-id. it is my database's view: 456 | DID | 14193016880 | 2 | hangup | | 455 | DID | 14193016880 | 1 | Dial | H323/1169#989181310524@66.152.61.66|60 | didx.org for

Estimado Jose: En nuestro caso utilizamos las mismas librerias y solemos resolverlo asi (donde el color depende del valor que se le asigne) //-------------------------------------------------------------------------------------------------------------------------------------- #configuramos donde setwd("D:/mapas/") #Carga de librerias library (sp) library (RColorBrewer) library

Dear for having an stable system which limit option is good for ulimit comand ? 2-is any option for making asterisk crash-free? Best -- Pezhman Lali -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110810/365d9d56/attachment.htm>

Dear, because of using dial(local/...) each incoming calls (_12X.) makes 4 ports on asterisk. I can not use goto , because of some limitations. is any way to decrease it? Best, [MAIN] exten => _12X.,Dial(LOCAL/${EXTEN}@TEST/n,60) .... [TEST] exten _X.,1,Dial(${EXTEN}@next_gateway,60)

Sorry about that, it was sent by accident. I have a data frame that looks something like this: id day k 56 -1 566 63 -1 680 73 -1 773 56 2 298 63 2 273 Of course, it is a very simplified version of the real data frame I am working on. I need to add another column that would represent a percent change in k from day -1, by id. I put only two ids at day 2 to

hi xten.de produced an activex for web phone. but I can not find any link for download. can u help me ? best Mani ____________________________________________________________________________________ Now that's room service! Choose from over 150,000 hotels in 45,000 destinations on Yahoo! Travel to find your fit. http://farechase.yahoo.com/promo-generic-14795097

dear can asterisk dial two numbers, then bridge them.(like jah jah) best Mani ____________________________________________________________________________________ Looking for earth-friendly autos? Browse Top Cars by "Green Rating" at Yahoo! Autos' Green Center. http://autos.yahoo.com/green_center/

hello: I am very confused when it comes to list operations in R. I seek help in the following problem. I have two different vectors myIDs - a character vector with no names to their elements x2 - another character vector derived from unlisting a list where an element has both a name and value. I am not happy the way x2 is with names and values to every element. > myIDs[1:10] [1]

hi@all I have the following problem: With asterisk 1.09 the grandstream's registers fine with both ports, with version 1.2.1 (the newest port on freebsd) I get "Unauthorized" SIP messages from the 2nd port. The ports are configured identically, the only difference is the sip and rtp port. On the first port the sip port is 5060 on the second 5062. The rtp on the first 5004 on the

Dear I am using Ser+Asterisk, for sip providing. there is a problem, the asterisk does not return back the busy tones to the sip phones. for example, if the destination number is busy, we are hearing waiting ring from sip phones, and after 60sec(timeout) the call will be terminated. thanks in advance for all help Best Mani __________________________________________________ Do You Yahoo!?

Hi now, I am using, realtime connection(mysql) for dialplan, but the following line must be added ,manualy to extensions.conf, before reloading.for each new context. [NEW_CONTEXT] switch => Realtime/@extensions is there any idea, to add this line to dbase too? thanks in advance Best MAni ____________________________________________________________________________________ Never miss an