[asterisk-users] sip trunk - SIP/2.0 488 Not Acceptable Media

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, I'm working on sip trunk between cme 4.x and asterisk (trixbox 1.2.3). Well, the trunk is partially working, asterisk' extensions talk with cme, but - - when cme try to connect to asterisk' number, receives the number dialed is not in

[Asterisk-Users] about * and CM/CME

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, on voip-info there's two howtos to connect asterisk with CM and/or CME Cisco, but always with sip trunk. What about h323 instead of sip? there's someone that has tested something like that? MWI will work too? Your feedbacks will be

[Asterisk-Users] about g729

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, my topology is like that: ISP --[sip]-- Asterisk --[sip]-- CME Cisco -- ip phones ISP services are g711 and g729 enabled. My Asterisk is registered on ISP with two sip UA. Then I've forwarded calls from ISP to ip phones registered on CME

[Asterisk-Users] problem with g729 and CME-Asterisk

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, my topology is: CME (Cisco) -- [sip trunk] -- Asterisk -- [sip trunk] -- ISP Services I need to connect my phones registered on CME to ISP Services using g729 codec. Well, on cisco I set the codec preference with a voice class: voice

Re: [Asterisk-Users] problem with g729 and CME-Asterisk

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Nov 9, 2005, at 4:33 PM, Greg Oliver wrote: Do a debug voip ccapi on the CME and look through it. It will have detailed codec negotiations, etc in it. thanks for your answer, Greg. Could you help me?

Re: [Asterisk-Users] problem with g729 and CME-Asterisk

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 I've forgotten my dial-peer config: dial-peer voice 500 voip description ext destination-pattern .T voice-class codec 1 session protocol sipv2 session target ipv4:192.168.17.10 dtmf-relay rtp-nte no vad 192.168.17.10 is *, .1 is CME.

Re: [Asterisk-Users] problem with g729 and CME-Asterisk

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Nov 9, 2005, at 5:18 PM, Greg Oliver wrote: Post up your dial-peer 500 config as well. It is doing codec 0x2 (g.711Alaw) from the get go. Also post relevant config for the phone from asterisk and dialplan entry used. the call flows are:

[Asterisk-Users] problem with CME on 12.3(11)T6 and later (MWI)

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, my topology: CME (cisco callmanager express 12.3(14)T4 on 1751v): 192.168.17.1 Asterisk (1.0.9 on Freebsd 5.4): 192.168.17.10 from 12.3(11)T6 and later CME sends num$ and not num only on sip trunks. See the 'sh sip reg status':

Re: [Asterisk-Users] problem with CME on 12.3(11)T6 and later (MWI)

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 mmm a solution is maybe something like that? http://mail.iptel.org/pipermail/serusers/2005-May/019677.html Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.4.1 (Darwin)

Re: [Asterisk-Users] problem with CME on 12.3(11)T6 and later (MWI)

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 this is from a sip debug on IOS: Nov 1 19:42:12 192.168.17.1 743: Nov 1 19:42:12 192.168.17.1 744: Nov 1 19:42:21 192.168.17.1 745: Nov 1 19:41:29.895 MET: //-1//SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr:

Re: [Asterisk-Users] about timeouts

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Maybe an example will be usefull: voip services -- * (5600) -- cme (601) -- * (5901) -- * (voicemail u601) voip services on sip.conf - - register = 530:[EMAIL PROTECTED]:5061/5600 [5600] type=friend

[Asterisk-Users] about timeouts

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, I've this infrastructure: |voip services| -- |*| -- |cme| -- |isdn| the voip services are logged on my *, then forwarded to number 601 on cme. The isdn calls too are forwarded to 601. On cme I've a timeout X for call-forward noan (no

[Asterisk-Users] about sip and skinny

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, it's possible to do something like that? x-lite -- sip -- [101 on sip.conf] -- skinny -- ccme [101 on callmanager] I hope that's clear enough. thanks for your advices Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4

[Asterisk-Users] about mpg123

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, how could I use rawplayer.c as http://www.voip-info.org/wiki-Asterisk+FreeBSD, or madplayer instead of mpg123? Thank you very much for your support Regards Andrea -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin)

Re: [Asterisk-Users] problems with call-forward from ccme to * on sip trunk

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Uhmm ... maybe a connection plar from ccme to an * number (like 511 on my conf), then a simple forward from 511 to 601 on ccme? Something like: exten = _511,1,Dial(SIP/601,45) I need help ... :D Andrea -BEGIN PGP SIGNATURE- Version: GnuPG

Re: [Asterisk-Users] problems with call-forward from ccme to * on sip trunk

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Apr 4, 2005, at 10:07 AM, Andrea Riela wrote: -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Uhmm ... maybe a connection plar from ccme to an * number (like 511 on my conf), then a simple forward from 511 to 601 on ccme? Something like: exten

Re: [Asterisk-Users] problems with call-forward from ccme to * on sip trunk

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 To help to find my mistake, I've two debugs: 1) isdn -- connection plar to 5600 on * -- 601 on cme -- vm call-forward to 5601 on * ext.num 123456789 calls my ISDN number, on ccme there's a connection plar to internal 5600 (on asterisk), that dials

Re: [Asterisk-Users] Asterisk Voice mail with CCM

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Apr 2, 2005, at 8:00 PM, Nathan Alberti wrote: I'm currently in the process of getting it to work for a CCME install, I have it all working except for one thing.. I think it was calling a phone from the asterisk server the call transfer back to

[Asterisk-Users] problems with call-forward from ccme to * on sip trunk

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks I've a strange problem, probably a mistake but I don't see it :( Description: My ephone-dn number on ccme, that is a simple connection plar for all ISDN calls, is 601 The voicemailmain on asterisk is 5900. CCME: 192.168.17.1 *: 192.168.17.10

Re: [Asterisk-Users] about sip and registering

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 On Mar 27, 2005, at 9:32 AM, Wilson Pickett wrote: [fwd-incoming] 123456,1,Dial(SIP/2000,45) Hi Wilson, thanks for your answer. The '2000' could be a local extension, or an external PBX extension (like a ccme number), isn't it? thank you very much

[Asterisk-Users] about sip and registering

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, I don't really understand the extension function on: register = user:secret:[EMAIL PROTECTED]:port/extension First question - -- well, it's only local, or is important for authentication on external sip server? Example: I've one

[Asterisk-Users] newbie questions

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Hi folks, I've some questions about asterisk, and in general about voip, please help me :) 1. I've SIP accounts on external servers, and I would that my local server will connect with those and redirect all calls from those to an internal SIP

[Asterisk-Users] about sip, asterisk and cisco ccme

for your support Regards dott. Andrea Riela -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.4 (Darwin) iD8DBQFCOLJtMakHrsrHP9wRAmDfAJ9AgcMf1CmdrLBk4HEdlvWKZiht7QCfcgns GbTX2LxGxO3ZR7iMIPqreJA= =eKlT -END PGP SIGNATURE- ___ Asterisk-Users mailing list

Re: [Asterisk-Users] problem with freebsd 4.9 port

On 17 Dec 2004, at 18:04, Jason Lixfeld wrote: Had the same issue with 5.3 1.0.2. Looks like some screwed up somewhere. well ... you think there's someone that could help us? thanks Andrea ___ Asterisk-Users mailing list [EMAIL PROTECTED]

[Asterisk-Users] problem with freebsd 4.9 port

Hi folks, when I try to compile the freebsd port (1.0.2) I see: ast_h323.cpp: In method `void MyH323Connection::SendUserInputTone(char, unsigned int)': ast_h323.cpp:725: argument passing to `const char *' from `char' lacks a cast ast_h323.cpp: In method `void