similar to: ULaw not negotiating

I'm using Asterisk 1.0.4 with AMP and Broad Voice. I have that with only 5 XTen Lite phones. I'm able to call / etc with internal phones just fine. I can call outside Vonage Numbers, and other BroadVoice Numbers. I have vonage where I live (626) and can call that fine. However, other 626 numbers I get similar errors as below. However, everytime, I try to call cell phones, and or

Hello, Have a weird problem when using asterisk (1.0.6). There are certain numbers I cannot dial when using asterisk with my broadvoice account. No problems with inbound. With outbound calls, I can call some numbers (for example broadvoice customer support number) and unsuccessfully with some. However, when I configure my account directly on x-lite, I dont see these outbound problems. Here is a

I can receive incoming calls. I can see via sip debug that I am communicating with BV just fine. When I call a BV number it goes through just fine. When I call any other # I get: "We're sorry your call can not be completed at this time. Please hang up and try your call again later" Broadvoice tech support does not see any errors and they see that I am registered just fine. I have

I have searched the list and cannot find a sip 400 solution posted that solves my problem. If anyone has any thoughts or suggestions on the following I would greatly appreciate it. I didn't have this error before Broadvoice made their changes this weekend. Now when I make a call it connects but, I cannot hear anything on the other end... The full message I have is: 8 headers, 0 lines

Hey, Sometime i am getting following messaged on asterisk CLI console just wondering what these messages are look like some codec related. [May 31 12:26:14] NOTICE[7349]: channel.c:4074 __ast_read: Dropping incompatible voice frame on DAHDI/i1/2031444389-28e of format slin since our native format has changed to 0x4 (ulaw) -------------- next part -------------- An HTML attachment

Sorry, I sent the following reply with the wrong "from" address and so it did not pass the lists spam filter. So here is the message again: Hi James, Thank you very much for your help! You were right, the codec is not compiled into my asterisk version. I'm using debian, too and a show translation tells me that it is deactivated. Would it be enought to only compile the ilbc codec

Hi: I have a mgcp.conf and a mgcp_additional.conf which records the special information about the extensions. And i found if i use ulaw in the general context in mgcp.conf,then all the registered extensions can make both outbound and inbound calls,the mgcp.conf is following: [general] port = 2727 bindaddr = 0.0.0.0 disallow=all allow=ulaw allow=alaw ; can be disable and do no effect

I'm running 1.4.22-rc5, and now see the following codec format listed:. What is 0x80004 (ulaw|h) ? 192.168.1.14 101 YjVlYzYwODd 00101/00002 0x4 (ulaw) No Rx: ACK 172.16.1.1 102 7b213e4762c 00102/00000 0x80004 (ulaw|h No Init: INVITE I'm having some voice quality problems and trying to see if it is related to this. Thanks Raja -------------- next part

Simple test extension exten => 14,1,Wait(1) exten => 14,2,SayPhonetic(${CALLERIDNAME}) exten => 14,3,Wait(1) exten => 14,4,SayDigits(${CALLERIDNUM}) exten => 14,5,Hangup Works fine from spa2k extension on lan Works fine calling broadvoice sip did When I call voicepulse sip did I get the calleridname and then silence. CDR logging looks okay but * messages log shows: Sep 19

Hello, I just CVS'd today and now I'm getting these errors when I call one grandstream phone to another both using 711U: NOTICE[1225991360]: File channel.c, Line 1476 (ast_set_read_format): Unable to find a path from ULAW to G723 NOTICE[1225991360]: File channel.c, Line 1446 (ast_set_write_format): Unable to find a path from G723 to ULAW NOTICE[1225991360]: File channel.c, Line 1476

Should I expect a standard fax machine connected to an ata-188 connected to an asterisk server, connected to a pri fed from a cisco 7206vxr to work correctly? It needs to have a standard fax machine, receiving and emailing it won't be acceptable. Thanks dave -- Dave Weis "I believe there are more instances of the abridgment djweis@sjdjweis.com of the freedom of the

Are there any considerations to take in account when faxing from analog to SIP using ULAW? The problem we're having is faxes are only making it halfway, getting cut off. Neither fax machine seems to report an error. Pretty diagram: FXS --> SIP --> PSTN Provider --> FAX ^ULAW

Greetings, I'm running latest * from CVS on FreeBSD 4.10 box. We've just bought several WellGate 3502A FXSes to play with till welltech guys fix the 3504a's registration bug. So far everything is working as expected, except the fact only ulaw and alaw codecs work with *. If I add allow=gsm or allow=g723.1 in FXS's ports entries in the sip.conf, no voice is heard from both

Ok, I'm quite fond of CVS-Stable 10-26-04 as it's always been fine. One thing I noticed with this version and all versions prior, when I did a "sip show channels" it always displayed info in all caps. But sometime between 10-26-04 and 12-8-04 they changed this to all lower case. I believe this MAY be related to the latest problem I just fixed. My provider was sending me

ulaw blank spots but gsm fine We've got plenty of QoSd bandwidth to run ulaw. Yet when we do, we get drops (blank spots) in the call. GSM codec eliminates the blanks, but is of course much less quality. Any ideas as to what would be causing this? Is ulaw less loss tolerant than gsm? I've run tests using iperf, and I'm looking at maybe 10 per 50000 packets are lost. (around 0.02%).

tuning for ulaw g.711 - Polycom IP500 I've got QoS ironed out at this point (dedicated section of bandwidth with VoIP having priority), yet ulaw is still unusable due to small 'holes' in the audio. Running iperf tests shows packet loss at very low percentages (< 0.02 %), and when we do drop, it's only 1 packet at a time. I thought ulaw was supposed to handle this OK.

Does anybody has got the some problem? The grandstream ATA 486 schould support almost all codecs, but it doesn't work. I get the following message when I force the use of different codec WARNING[9529]: chan_sip.c:2765 process_sdp: No compatible codecs! Feb 3 11:17:15 NOTICE[9529]: chan_sip.c:7395 handle_request: Unable to create/find channel What could I do to see some more detailed

For some reason a couple weeks ago users began experiencing garbled audio in one direction when dialing out via our VoIP provider. This happened at multiple sites simultaneously. The VoIP provider doesn't think it's their problem. If I switch to another codec so that Asterisk transcodes everything is fine. On conference calls (where Asterisk gets in the middle to relay ulaw to all

Hello, I'd like to use g729 pass-thru when I dial out to a sip provider from my IP phone but because I have no license for g729 I'd like to use g711 ulaw for asterisk voicemail, conference bridge and other services. When I set in [general] section of sip.conf the following: disalow=all allow=g729 allow=ulaw the g279 pass-thru works fine with my SIP provider but when I call the

Mar 15 12:54:01 NOTICE[24269] channel.c: Dropping incompatible voice frame on Local/[removed number]@context-5c3e,2 of format ulaw since our native format has changed to slin Can anyone provide an English translation of what this means? The extension is a Polycom IP 501 The only allowed formats are g.711u MOH is MP3 files (obvious) All prompts have been re-recorded in .ul uLaw