similar to: SetCIDName

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi. Using inAccess Networks chan_oh323, I'm experiencing some clicks or pops, how can I fix that? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/QMfF2TEAILET3McRAu9zAJwNWtv+QSpka0NGaVk9E/IDHyalhwCgkQME Gynfp5zF0SWZUQEjelp7sBI= =CSqT

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Now that I've been unable to register 2 hardware SIP phones and one software (Kphone), I'm beginning to doubt that chan_sip works at all. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux)

Hi, Does Asterisk support SS7? Google shows an old new post from Feb. 2002 stating that OpenSS7 would help add SS7 support to Asterisk, but presently OpenSS7 seems to be dead and I can't seem to find anything about it at Asterisk or Digium's sites. What happened? -- Regards, Tais M. Hansen ComX

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm having a problem with the following expression examples. exten => s,1,NoOp($[$[${value} >= 10] & $[${value} < 18]]) exten => s,1,GotoIf($[$[${value} >= 10] & $[${value} < 18]]?3) ${value} is 13 in both examples above. First extension evaluates to 1 while second evaluates to 0 even though it's the same

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, After upgrading the kernel on an Asterisk box, asterisk segfaults on startup. It seems like it's the g729 codec that causes this: #0 0x4015acad in memset () from /lib/libc.so.6 #1 0x4022686a in load_module () at codec_g729b.c:416 #2 0x08054794 in ast_load_resource (resource_name=0x80d1068 "codec_g729b.so") at loader.c:298 #3

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I was just measuring the bitrates of a couple of codecs via iax. I'm getting much higher numbers than expected, so maybe I'm doing something wrong? Measured with iptraf, values displayed are: codec: measured bitrate (bitrate according codec definition) gsm: 52 kbps (13 kpbs) alaw: 154 kbps (?) speex: 57 kpbs (24 kpbs) Seems a little

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, No matter what I configure my spans at (on a E400P) ztcfg -v always shows: SPAN x: D4/ AMI Build-out: 0 db (CSU)/0-133 feet (DSX-1) Currently I've configured my spans as "ccs,hdb3,crc4", so shouldn't D4/AMI be showing "ccs/hdb3" instead? - -- Regards, Tais M. Hansen ComX -----BEGIN PGP SIGNATURE----- Version:

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, When congestion doesn't do anything on a zaptel channel, would it be because the tones in the zaptel zonedata.c isn't correctly defined? -- Nobody picked up in 15000 ms -- Executing Congestion("Zap/120-1", "") in new stack ... But the calling phone just keeps ringing. - -- Regards, Tais M. Hansen ComX

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Before I hack out the ',' -> '|' tr in extension.conf parser, any way to escape ',' that I missed? - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/ZfuP2TEAILET3McRAjXUAJ0VjuFeABe5jqpSlrBakDC2IMjvrQCfcBYU

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm attempting to hook up the E400P card to a Siemens ISDN module. I have no knowledge of these Siemens products, so I'm acting on what I've been told about it. The Siemens side is configured to "ISDN30: ECMA QSIG". The Siemens manual states the card provides 30 ISDN B-channels which can be used for trunking or

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I've been unable to get app_transfer to work, could someone explain how it is supposed to work? Currently I have two Asterisk boxes. A call comes in via zaptel to ast1. ast1 dials ast2 using iax2 and gets instructed to transfer the call to a different extension. iax2 debug shows that a transfer cmd is sent to ast1, but nothing happens

Just updated from CVS 12-23-03 to tarbal 0.7.1. Identical settings in zconfig.h for echo cancellation (MARK2, aggressive OFF). The echo got worse, much worse. It takes longer to train and overspeak now disables cancellation, which it did not before. In fact, I now have echo on VoIP to VoIP on our local network, which I never had before. Was something changed with the echo supression which

Considering that I had a failure with compiling the latest version of asterisk because of cdr_mysql, I am going to assume that I need to have a copy of the mysql headers on the system in order to compile cdr_mysql.so. Does that sound correct? Kim C. Callis -------------- next part -------------- An HTML attachment was scrubbed... URL:

Good Afternoon Everyone, I am having a problem with compiling the CVS version of *-addons downloaded today. I am also having problems compiling an older version as well but im ignoring that one for now. I believe I have all the correct libraries, and I have done extensive searches everywhere I just wondered if I was missing something really silly, or if this is a problem other people have

Hello- I'm using the "accountcode" field in cdr table to store some temporary information used by my perl scripts. I'm finding that the field seems to be truncated to 19 characters, even though the allocated field in the table is larger (varchar(45)). Anyone familiar with the asterisk sources know where I can make a change to keep this from happening? ie: to allow more

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I understand it, IAX2 trunking requires zaptel timing. Zaptel timing is provided by Zaptel cards, ztdummy or ztrtc. ztdummy requires usb-uhci and ztrtc can't run on smp systems. So if you only have smp systems with ohci and no zaptel cards (because it's a sip/iax2 gateway) then you're screwed? - -- Regards, Tais M. Hansen

Hi, Does Asterisk have a limit to how many NAT'ed SIP clients it supports behind a single IP? I have the weirdest problem ever. I have three SIP endpoints. SNOM phones, if it matters. Their extensions are 200, 201 and 202. Apart from the username/password, the sip entries in sip.conf all have identical configuration. They're all NAT'ed behind the same IP. 200 and 202 registers

Hello, is it possible to change how are caller id on incoming call from isdn, capi lines displayed od sip phones ? ( e.g. SNOM ) standard is 1234567@domain.net. I just want only 1234567 to be displayed. is it possible ? regards Marian -- SUNTEQ s. r. o. Hviezdoslavova 9 # Prievidza # 971 04 # Slovak republic Tel: +421-46-5430 754 # Fax: +421-46-5439 144 http://www.sunteq.sk/

hi .. maybe someone can help me, I seem to have lost the sheet of paper that comes with an E100P card and tells you how to compile the stuff it requires to run. I'm trying to move my Asterisk to a different box and at this time totally stuck. Could someone be kind enough as to mail me a PDF of it ?? many thanks Dave

Hello. I'm seeing this at the console. NOTICE[81926]: File chan_sip.c, Line 5119 (handle_request): Registration from '<sip:marrandy@192.168.1.1>' failed for '192.168.1.70' What's this all about ? Regards...Martin -- Osborn's Law: Variables won't; constants aren't.