similar to: Asterisk timing

-----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)

-----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'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, 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 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, 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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, As I understand it, Asterisk currently uses the timestamps in incoming RTP packets to build outgoing voice frames. Is this true? Would it be possible for me to use i.e. zaprtc as a timing source for the outgoing stream? I.e. in a setup like below I'd like to use zaprtc timing on Ast1 because I don't trust the timestamps coming from

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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, How can I check if (i.e.) my provider is requesting me to hide the callerid? I.e. (Telco)E1/PRI---Zap(Asterisk1)IAX---IAX(Asterisk2)SIP---EP Now, if a call comes from the Telco with CLI screening prohobited to Asterisk1, where the call is forwarded using Dial() via IAX to Asterisk2 and then (also using Dial()) on to a SIP endpoint, how do I

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Seems like Asterisk/chan_sip in some special cases gets it's rtp channels mixed up. I've got a few reports on users hearing someone elses conversation on the line. Could be port problems, but I haven't had time to make any traces or tests yet. Before I start to analyse this periodic problem, I thought I'd just check with the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, When making a call like the one below, I get double DTMF tones on the PSTN side. DTMF tones sent from the PSTN arrives squelched on the SIP side. SIP > Asterisk2 > IAX2 > Asterisk1 > ZAP > PSTN SIP has been configured to use rfc2833 on both the SIP endpoint and the Asterisk. SIP endpoint also suggests a payload value of 101.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, Any reason why pickupgroup has been limited to 31? 31 groups are quickly used up when you have multiple companies on the same server. - -- Regards, Tais M. Hansen ComX Networks Tel: +45-70257474 Fax: +45-70257374 -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.3 (GNU/Linux) iD8DBQFBEOMg32si/nlrQ5gRAu3+AJ9FkeGMgb1JaAy2WjY8wBNEsN4WnwCeMFP0

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

Yes, I am a newbie too. I am having a problem with meetme. From what I have seen it will work without a Digium card but with audio problems. My goal is just to see how it works not the quality of the audio. When I dial into the conference room the following message is played: "That is not a valid conference number." On the console I get: "unable to open pseudo channel". As

-----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, Why do the string specified with SetCIDName() get overwritten with the name of the user running Asterisk, when placing calls through H323? - -- Regards, Tais M. Hansen ComX -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.2.2 (GNU/Linux) iD8DBQE/J5YL2TEAILET3McRAuFUAKCiLlSXB4qzEVcm8ps8VHzkGlPLewCeLdig x399jPG6fIEvyxf52ksi/nQ= =dhvu -----END

-----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, 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

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