search for: asteirsk

Has anyone managed to get gtalktovoip working at all? If so please explain. http://www.gtalk2voip.com/faq.shtml 2. Q: Ok, how can I call Google Talk, MSN or Yahoo users from SIP ? A: This is a major feature of our gateway and it is very easy. o GTalk: user@domain.com can be reached by calling to sip:user_at_domain.com@gtalk.gtalk2voip.com o MSN: user@domain.com can be

...roblem appear,when i dial the number into the pbx,sometimes i can not listen to the ivr ,and no rtp create. if i unload the ztdummy driver,the proble will disapper. I guess may be the timer source problem, but i dont't know how to solve it . anyone can give me some advices will be appreciated. asteirsk-1.4.21 and zaptel-1.4.10 Thanks in advance! -- Best regards! jordan pan Location:Shenzhen China Company:www.justcall.cn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100712/2b6af190/attachment.htm

Hi, Is there any way of telling Asteirsk to withhold answer subversion on a call till I call Answer. My DP looks like this: [incoming] Exten => 18005551212,1,Noop() same => n,Answer same => n,Mset(__uid=${SIPCALLID}) same => n,MixMonitor(/tmp/FROM_CALLER_${uid}-${START}.WAV) same => n,Dial(Local/1 at dial_call_center/...

I have a setup of asterisk 1.6 in one box and asteirsk 1.4 in another. I can send recieve faxes from both boxes fine to and from pstn. But the faxing between 1.6 and 1.4 extensions does fail. Any ideas please ? -- Thank You Amit Nepal

hello, i got this error when i run make after downloading asteirsk from cvs. gcc -pipe -Wall -Wstrict-prototypes -Wmissing-prototypes -Wmissing-declarations -g -Iinclude -I../include -Iinclude/solaris-compat -I/usr/local/ssl/include -D_REENTRANT -D_GNU_SOURCE -O6 -Wcast-align -DSOLARIS -DBUSYDETECT_MARTIN -fomit-frame-pointer -c -o term.o term.c...

..._odbc is working but i hav problem in res_odbc. i have created user in sip_buddies table but asterisk is no getting user from this sip_buddies table. /etc/asterisk/extconfig.conf [settings] sipusers=>odbc,asterisk,sip_buddies sippeers=>odbc,asterisk,sip_buddies /etc/asterisk/res_odbc.conf [asteirsk] dsn=>asteriskdsn username=>voipbilling password=>voipbilling pre-connect=>yes __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

Question is quite easy: How am I supposed to configure Asteirsk to have the same extension, in 2 differents domains. In the general section of sip.conf, I add the domains, But how to say to Asterisk : user1@domain1 > Pasword1 user2@domain2 > Pasword2 Thanks for your help !!!!! JM -------------- next part -------------- An HTML attachment was scrubbed....

I've got a PAP2 that I've got working with asterisk. At the moment, its configured so that when a phone is picked up on it, it connects to Asterisk. My hope is that I can let Asteirsk handle the entire dialplan, including dial tone generation. What would my context in extenstions.conf look like for this sort of dialing. More accurately, how can I get Asterisk to generate the dial tone on the pap2's line on connect (holding the dial tone past the initial 9, dropping it with a...

Hi, can i set up my asterisk for: - receive a skinny call in a specific context (yes, i have already compiled asteirsk with h323 support) - forward the call to a sip user A - make the sip user B join the call and create a conference between skinny caller, A and B maky thanks

...ave read the wiki and several other internet documents. Can anyone make a comment as to what kind of functionality will you loose if you use Cisco 7940 phones with asterisk 1.4 things like: MWI, call transfer, conference,etc,etc. I have a customer with 6 of those phones that he like to use with the asteirsk PBX. thanks, -- ------------------------------------------------------------ Erick Perez ------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/2007...

http://gizmodo.com/gadgets/cellphones/sprintsamsung-instant-cell+to+wi+fi-box-is-official-named-airave-300451.php The above is quite interesting, it would be interesting to see if it uses sip, which I have no reason to believe otherwise, and if it does, can it be hacked to talk to Asteirsk? In which case one could have a very good extension to asterisk using any Sprint Cell phone, or maybe even any CDMA (Verizon) cell phone as well.

1.4 was working fine. I thought I would try 1.6 beta 9 from my asteirsk 1.4 server to my asterisk client 1.6beta it wont accept the call. [Jul 18 20:34:55] NOTICE[966]: chan_sip.c:16416 handle_request_invite: Call from 'JJ' to extension 'jj_audio' rejected because extension not found. I changed nothing in the config files. I tried setting debug lev...

Hi all, In sip.conf: [general] regcontext = autoreg [devabc] regexten = 543 creates "exten=> 543,1,Noop(devabc)" in context autoreg when devabc registers. But I can't use "exten=> _5XX,2,Dial(SIP/${EXTEN})" in the dialplan, because there's no device SIP/543. Now I know I can add a line like "exten=> 543,2,Dial(SIP/devabc)" for each and

Hello, I am posting this after spending hours digging through the list archives. Problem : When asteirsk plays a voice prompt, the voice clip is really choppy. I figure that this is something to with the sound card, the timing of playback etc. But cannot seems to find an answer. Here is the Notice which appear when voice prompt is played. NOTICE[1217602880]: File sched.c, Line 209 (sched_settime):...

...build_tools/menuselect-deps config.status: creating makeopts Sincerely, K -----Original Message----- From: KC [mailto:mrprotocols@gmail.com] Sent: Friday, March 30, 2007 1:43 AM To: 'asterisk-users@lists.digium.com' Subject: Asterisk-Addon-1.4.0 MySQL I can't find anything about Asteirsk-Addon-1.4 MYSQL problem from googling. I thought it would be my error but surely not just tried asterisk 1.2.17 with addon 1.2.5 and it work. Does anyone else having problem to make res_config_mysql, cdr_addon_mysql and app_addon_sql_mysql in addon-1.4? Thanks for sharing There are no res_config_m...

...00002 [192.168.22.251:4569] Oct 2 10:05:41 NOTICE[32273]: chan_iax2.c:2880 auto_congest: Auto-congesting call due to slow response -- IAX2/192.168.42.250:4569-4 is circuit-busy Tx-Frame Retry[000] -- OSeqno: 001 ISeqno: 000 Type: IAX Subclass: HANGUP On the receiving Server Asteirsk 1.4.x Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass: AUTHREQ Timestamp: 00003ms SCall: 00657 DCall: 00002 [192.168.25.250:4569] AUTHMETHODS : 3 CHALLENGE : 152361611 USERNAME : priv Rx-Frame Retry[ No] -- OSeqno: 000 ISeqno: 000 Type:...

...dio works in one direction only for the asterisk (SIP) - gtalk call. anyone else got asterisk - googletalk using chan_gtalk working? > >Message: 10 >Date: Fri, 02 Mar 2007 19:07:41 +0200 >From: Cosmin Prund <cosmin@adicomsoft.ro> >Subject: Re: [asterisk-users] gtalktovoip and Asteirsk >To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> >Message-ID: <45E859DD.8080103@adicomsoft.ro> >Content-Type: text/plain; charset="iso-8859-1" > >I don't think it works. I tried calling my own yahoo messenge...

Hello Any one using asterisk-prepaid with mysql. i want asteirsk-prepaid for fedora core 2. i have installed mysql-devel. but after that i am unable to compile the asterisk-prepaid it is giving me error for libmysqlclient. i already have this library in my /usr/lib/mysql. i am using asterisk-CVS. Here is the error given when i try to compile asterisk-prepaid....

Hi there. The title basicly explains it. When we get incomming calls from cellular phones, the callers tend to echo ALOT. They hear their own voice at very high volums. This is a problem only for mobilphone users that calls in to us. I'm using wifi IP-phones. Asteirsk: CVS-Nv1-0-7-04/19/05 Any way to fix this on the asterisk server? Regards, Arne Morten -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050901/b91a67d6/attachment.htm