search for: 711

hi, i have strange problem with no rtp packets from asterisk after dns query. see pcap below centos6/asterisk 13.9 + chan_sip 172.23.0.3 - asterisk 172.23.5.1/2 - voip phones any ideas/hints? 1170 25.028206000 172.23.0.3 -> 172.23.5.1 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x334508F6, Seq=49318, Time=1442112256 1171 25.045556000 172.23.5.1 -> 172.23.0.3 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x643C9869, Seq=4468, Time=716240 1172 25.045629000 172.23.0.3 -> 172.23.5.2 RTP 214 PT=ITU-T G.711 PCMA, SSRC=0x3566361, Seq=60990, Time=716240 1173 25.04...

Audio packets are running... 961 16.150421076 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28402, Time=73280 962 16.170411284 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28403, Time=73440 963 16.190381989 192.168.5.150 -> 192.168.5.25 RTP 214 PT=ITU-T G.711 PCMU, SSRC=0x6A3E0AF1, Seq=28404, Time=73600 964 16.210387990 1...

I posted this question a while back, and I'm posting again in hopes that someone has some ideas. Sorry if you've already seen this. When dialing out using a SIP or IAX provider (Broadvoice, SimpleTelecom, VoicePulse Connect) I often find that after the call is answered the first few seconds of audio are cut off (i.e. I don't hear the called party). This usually results in the called

Hello list, We're currently facing some issues concerning T.38 gateway faxing. This is a device used almost exclusively for receiving faxes. Calls are incoming to asterisk on a SIP trunk (sangoma netborder) using G711A. Gateway mode is activated in the asterisk dialplan towards a Cisco SPA 112 running firmware 1.3.5. We are using asterisk 1.8.13.0 with the T.38 gateway patch applied (I know I know, we should move to asterisk 11. I'm trying that tonight after business hours). The issue we're seeing is th...

Hi folks. The jspeex library has classes for converting speex to pcm and vice-versa. I also have other code that converts from G.711 to pcm (and vice-versa). I want to transcode G.711 to speex, using an input stream. Can I accomplish this in one step, or must I go G.711 -> PCM -> Speex? If possible in one step, is there some example code I could look at for reference? Thanks! -- Jeff Ramin Software Engineer Singlewir...

*This message was transferred with a trial version of CommuniGate(tm) Pro* Will asterisk actually convert between two different codecs????? ie, a SIP endpoint running GSM and another running G.711? Wouldn't that add quite some latency? I was always under the impression Asterisk did not recompress and was smart enough to negotiate the right codec at each end and just pass through the RTP packets. Regards, Jamie Carl Email: me@jazz-inc.net PH: +61-414-365-466 -----Original Message---...

Here is the scenario... SIP UA's can use either GSM or G.711 ( in that order of preference in the sip.conf ).. Asterisk Server1 is linked to Asterisk Server2 via IAX2 and also supports GSM and G.711 ( also in that order of preference).. 1. If a call comes in from the UA using GSM and then goes out over the IAX2 leg, Will Asterisk simply move the GSM enco...

Using FireFly, all other codecs but G711 Ulaw is selected. But whenever I place a call, I get: Jan 24 16:07:06 WARNING[30654495]: dsp.c:1468 ast_dsp_process: Inband DTMF is not supported on codec G.711 u-law. Use RFC2833 Umm, wtf? I thought Inband was ONLY supported on G.711 u-law. -------------- next part -------------- An H...

Does anyone have the C reference code of the ITU G.711 Appendix II ? -- Guilherme Loch G?es "Wave after wave will flow with the tide And bury the world as it does Tide after tide will flow and recede Leaving life to go on as it was..." - Neil Peart , Natural Science

Hello, I have an asterisk server running with 23 g.729 licenses. I have also purchased a sound file from thevoice.digium.com. I need to covert this file (uLaw, PCM I think) to g.711, g.729 & g.723 for use with an IVR system. Is there a way I can convert the files using the g.729 digium codec? sox? Thanks -Matt -- Matthew S. Crocker Vice President Crocker Communications, Inc. Internet Division PO BOX 710 Greenfield, MA 01302-0710 http://www.crocker.com

http://bugzilla.netfilter.org/show_bug.cgi?id=711 Summary: iptables -m iprange causes unknown error Product: netfilter/iptables Version: linux-2.6.x Platform: All OS/Version: All Status: NEW Severity: normal Priority: P5 Component: ip_tables (kernel)...

Hi All, For a small installation using ITSPs via DSL is G.729 a worthwhile exercise? I have G.729 capable SIP phones and my ITSPs cupport the codec so I could go end-to-end without transcoding. What's call quality like compared to G.711, GSM or iLBC? Michael -- Michael Graves mgraves@pixelpower.com Sr. Product Specialist www.pixelpower.com Pixel Power Inc. mgraves@mstvp.com o713-861-4005 o800-905-6412 c713-201-1262

...calling host 3500@192.168.1.150:1720 -- Call token is ip$localhost/29426 -- Call reference is 29426 -- DTMF Payload is [pt=101] -- Called 3500@192.168.1.150 Setting capabilities to 0x8 (alaw) Capabilities in preference order is (alaw) Allowed Codecs: Table: G.711-ALaw-64k <1> UserInput/hookflash <2> UserInput/RFC2833 <3> UserInput/dtmf <4> Set: 0: 0: G.711-ALaw-64k <1> 1: UserInput/hookflash <2> 2: UserInput/RFC2833 <3> UserInput/dtmf <4> -- Send...

Okay this may be a very noobie question, but I've looked around the WIKI and haven't found anything about this. I was looking at some VO-IP phones, an they list one of the codecs as g.711, this isn't listed on the * page I was looking at, is this also known as another name, or is it not supported by *? Thanks Joel Duffield Near North Business Machines www.NearNorthBusiness.com

...r/share/ssl/certs/ldap.pem TLSCertificateFile /usr/share/ssl/certs/ldap.pem TLSCertificateKeyFile /usr/share/ssl/certs/ldap.pem I restart the service. Then, I run the comando authconfig and I select ldap with tls. I review the logs ldap server a thrown the following: Mar 5 11:54:38 eucalipto slapd[711]: conn=13 fd=14 ACCEPT from IP= 172.16.12.160:33935 (IP=0.0.0.0:389) Mar 5 11:54:38 eucalipto slapd[711]: conn=13 op=0 STARTTLS Mar 5 11:54:38 eucalipto slapd[711]: conn=13 op=0 RESULT oid= err=0 text= Mar 5 11:54:39 eucalipto slapd[711]: conn=13 fd=14 closed (TLS negotiation failure) I need you...

...patch .copfile {border:1px solid #ccc;margin:10px 0;} #patch ins {background:#dfd;text-decoration:none;display:block;padding:0 10px;} #patch del {background:#fdd;text-decoration:none;display:block;padding:0 10px;} #patch .lines, .info {color:#888;background:#fff;} --></style> <title>[711] trunk/wxruby2/Changelog: Ruby extension directory, paint fix, multi_line_text_extent, doc publishing</title> </head> <body> <div id="msg"> <dl> <dt>Revision</dt> <dd>711</dd> <dt>Author</dt> <dd>brokentoy</dd>...

...you can try to increase the size of the ==29930==? main thread stack using the --main-stacksize= flag. ==29930==? The main thread stack size used in this run was 8388608. ==29930== 64 bytes in 1 blocks are possibly lost in loss record 35 of 77 ==29930==??? at 0x4C2B9B5: calloc (vg_replace_malloc.c:711) ==29930==??? by 0x5277AB5: pool_system_malloc (mempool-system.c:75) ==29930==??? by 0x525D2FD: p_malloc (mempool.h:99) ==29930==??? by 0x525D2FD: hash_table_create (hash.c:70) ==29930==??? by 0x51F377C: settings_parser_init_list (settings-parser.c:215) ==29930==??? by 0x51ED102: master_service_set...

Dunno why your phone isn't allowing you do negotiate g711u but I can tell you how to upgrade the firmware. I called them on Thursday for myself and they gave me the following tftp server address for which to program my phone. 4.3.153.50 Load this into your phone's tftp area and reboot it. It'll go out to the net and check the firmware revision a...

http://bugzilla.mindrot.org/show_bug.cgi?id=711 Summary: 3.7.1p2 does not compile on redhat 5.1 Product: Portable OpenSSH Version: -current Platform: All OS/Version: Linux Status: NEW Severity: normal Priority: P2 Component: Build system Assigned...

Has anyone done anything with Asterisk using the G.711 codec? Also, is there a uncompressed option so that you could assign a single port to be unconpressed audio? Thanks! Stu -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030602/118e6ac7/attachment.htm