similar to: SIP/IAX2 DTMF

-----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, 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 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, 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 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, 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, 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

-----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, 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

-----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

In the trace below, swiotlb_alloc() is called with __GFP_NOWARN, it ors attrs with DMA_ATTR_NO_WARN and passes it to swiotlb_alloc_buffer(), which does NOT pass it on to swiotlb_tbl_map_single(), leading to an ever repeating warning that the caller of swiotlb_alloc() explicitly asked to be squelched. Pass the caller's request for silence onward. Xorg-3170 [006] .... 963.866098:

Jean-Marc Valin wrote: > Andy Ross wrote: > > Uh, production applications almost always require squelch, no? > > Some do, some don't. In general, distinguishing between a keyboard > and a speech transient is next to impossible based only on a few ms > of speech. That is true for distinguishing it by waveform, but not by amplitude. As I mentioned, these transients are

Hi again, Well - I didn't see beta8a-2.3.3 in custom dir. Will try. Also I tried to contact Sangoma - they are very fast to answer but main problem is time difference - it's 6 hours between Canada and Europe. Br, dmitry Dmitry Zhukovski System developer ComX Networks A/S Naverland 31, 2 DK-2600 Glostrup Denmark Phone: +45 70 25 74 74 Fax:???? +45 70 25 73 74 Web: www.comx.dk

I've written an Ogg Vorbis (only !) player, which some may like to play with. It works for me (tm) and I like it. If you don't, well, you know what you can do with it ;) I leave it running 24/7 and it plays my music without problems and without annoying me. That's all it's for, really. You can get it from http://www.geoid.clara.net/rik/arch/squelch.tar.gz The README follows ...

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'm trying to connect an E400P card to a Siemens Digital ISDN S2M Module. (TMS2). How should I connect these two? Straight or crossed? (receive tip <-> receive tip or receive tip <-> transmit tip) Also, I can't seem to figure out how to configure the channels. Does anyone here have any experience with the Siemens module

-----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, I'm having problems with hungry h323 channels. I think I've seen a question about this problem on this list before, but I couldn't seem to find it again. The problem is that the first 2-3 seconds of audio after Answer from Asterisk to an h323 phone disappears in thin air. The easy (but ugly) way to solve this is to add a 3 second