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Hi: Iam looking for a sip softphone that supports g729 codec Any one have an idea ? Reagrds; jonnyhashem --------------------------------- Don't get soaked. Take a quick peak at the forecast with theYahoo! Search weather shortcut. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070702/b0ad2034/a...

How suitable do the developers think that LLVM would be as a code-generator for a dynamically typed langage? Would the lack of static type information make the traditional code optimizations performed by LLVM relatively ineffective? Sincerely, Rayiner Hashem

i have added port=5060 to sip client configuration but it seems the same problem and in the same errors: Aug 11 10:29:18 WARNING[9869]: chan_sip.c:843 retrans_pkt: Maximum retries exceeded on call 04b3ccd87e45e719588c54a4017e3b99@172.16.180.21 for seqno 102 (Non-critical Response) __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam

HI: what is command on console to know how many calls are sending in the same time? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

when i send calls from an asterisk box to a voip provider the call fails and give me these messages: *CLI> Sep 23 21:32:32 WARNING[14595]: chan_sip.c:6890 handle_response: Forbidden - wrong password on authentication for INVITE to '"asterisk" <sip:asterisk@195.112.214.99:5070>;tag=as19e688a1' -- SIP/call-0f60 is circuit-busy == Everyone is busy/congested at this

...h the dispatch mechanism is different. Since llvm doesn't appear to have any notion of inheritence, C++/Java virtual methods should need to cast their 'this' pointer to the precise type. I presume this doesn't interfere too much with the other optimizations? Sincerely, Rayiner Hashem

02.11.2023 18:04, MATYAS, Tibor via samba: > Dear all, > > updating openssl from 1.1.x to 3.x on our gentoo systems (recompiled everything against the new openssl!) > breaks kinit: FWIW, most distributions switched to openssl3 quite some time ago. Eg, current libssl in Debian is of version 3.0.11-1~deb12u1. You did not provide any other useful info. Two of the most important

...rb5?? And what's your DC? > > /mjt > -------------------------------------------------- DSI Aerospace GmbH Sitz der Gesellschaft: Otto-Lilienthal-Str. 1, D-28199 Bremen, Germany Web: http://www.dsi-as.de Geschaeftsfuehrer: Dr.-Ing. Christian Dierker M. Sc. Elias Hashem Handelsregister: HRB 17726, Amtsgericht Bremen Umsatzsteuer-Identifikationsnummer: DE 192 681 774 --------------------------------------------------

On Thu, 11 Sep 2003, Rayiner Hashem wrote: > How suitable do the developers think that LLVM would be as a code-generator > for a dynamically typed langage? Would the lack of static type information > make the traditional code optimizations performed by LLVM relatively > ineffective? LLVM is designed to be able to suppor...

...olution found so far. Thanks in advance, Tibor -------------------------------------------------- DSI Aerospace GmbH Sitz der Gesellschaft: Otto-Lilienthal-Str. 1, D-28199 Bremen, Germany Web: http://www.dsi-as.de Geschaeftsfuehrer: Dr.-Ing. Christian Dierker M. Sc. Elias Hashem Handelsregister: HRB 17726, Amtsgericht Bremen Umsatzsteuer-Identifikationsnummer: DE 192 681 774 --------------------------------------------------

On Thu, 2023-11-02 at 16:04 +0100, MATYAS, Tibor via samba wrote: > Dear all, > > updating openssl from 1.1.x to 3.x on our gentoo systems (recompiled > everything against the new openssl!) > breaks kinit: > > kinit administrator at xxxx > administrator at xxxx's Password: > kinit: rc4 8: EVP_CipherInit_ex einit > > kinit -F -k -t /etc/dhcpduser.keytab -c

It does not look like Nufone is still in business, judging from the content on their site, which is very little. There is not even a configuration document to download, to connect to their network. The rates file is only for US/Canada calling. No international rates on this rates.csv file. I have signed up with a $5.00 account with them way back in November 2004. After signup, I havent received

i had a sip invitation problem with my voip provider and here the message that was shown : Oct 16 20:23:19 WARNING[21901]: chan_sip.c:6890 handle_response: Forbidden - wrong password on authentication for INVITE to '"asterisk" <sip:XXXXX@195.112.214.99>;tag=as7b43dfbd' -- SIP/callshop-3fcc is circuit-busy == Everyone is busy/congested at this time -- Got SIP

HI: Is there any way to raise up sound volume on fxo on TDM04B without changing tx-gain and rx-gain ? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

With regard to Bartlett-style collectors, they are also used in CMUCL/SBCL, OpenDylan, and various products based on Ravenbrook's Memory Pool System. Also, while Mono doesn't use a Bartlett-style collector, it does support pinning objects referenced from ambiguous stack roots, in an otherwise copying collector. While ITA has expressed concerns with SBCL's GC, they seem to involve the

when i display in extensions.conf this : exten => s,1,Answer exten => s,n,DeadAGI(astcc.agi,${CALLERIDNUM}) exten => s,n,hangup when a call comes the zap doesnot rerad the callerid an give me this: Jul 15 10:45:24 WARNING[6693]: chan_zap.c:5739 ss_thread: CallerID returned with error on channel 'Zap/4-1' excuse me for my english iam italian ,thanks in advance.

does anyone had an experience with not hanging up the call in astcc, thats my problem sometimes call does not hang up automatically and even when pressing star the call continue without stopping. Regards; jonny ____________________________________________________ Start your day with Yahoo! - make it your home page http://www.yahoo.com/r/hs

i have opened an account with callshopcompany,and when ive tried to send calls by the sip i had a message show an asterisk invitation problem i had these sip configuration: sip.conf [callshop] type=peer host=sip.callshopcompany.com username=XXXXX secret=XXXXXXX Then i tried to add these lines and it worked : sip.conf [callshop]

after editing the Make file i tried to compile asterisk-oh323-0.6.5 but many errors displayed like this: [root@localhost asterisk-oh323-0.6.5]# make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/home/ali/asterisk-oh323-0.6.5/wrapper' ./check_ver /home/ali/pwlib pwlib ./check_ver /home/ali/openh323 openh323 ar rc liboh323wrap_s.a

after editing the Makefile according to my directories ive compiled asterisk-oh323-0.6.5 but i got these errors can any body help me with this : [root@terra@net asterisk-oh323-0.6.5]# make for x in wrapper asterisk-driver; do make -C $x build || exit 1 ; done make[1]: Entering directory `/home/wassim/asterisk-oh323-0.6.5/wrapper' ./check_ver /home/wassim/pwlib pwlib ./check_ver