similar to: Does Early Media have to be Ulaw?

We're dialing a disconnected number via Level 3's vector network, and are receiving this. The response has SDP in it. Apparently, Level 3 is playing early media. Asterisk doesn't seem to know what to do with SDP in a 180 RINGING, and just plays ringing. What am I missing here? How can Asterisk see there's SDP, early media, in the response and act accordingly? SIP/2.0 180

Hiya, I sent this bugfix to the asterisk-dev mailing list, and modified it as I noticed side effects, but now it appears to be finished. Nobody seemed to notice it there, so I thought I'd post here, as it seems to be something that will be needed as people update to the latest CVS version. So...read on :) Ted programmer_ted@hotmail.com P.S. Read to the very end. The original bugfix

I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1]

anyone has idea what problem can be here, something with codec but i have today CVS version and grandstream phone with 1.5.0 firmware.I try to change codec in phone and also in asterisk-sip.conf but the same. What can be problem ? tnx, Tomaz *CLI> -- Executing Dial("SIP/102-767c", "CAPI/2:5") in new stack -- Called 2:5 -- CAPI[contr1/2003002]/0 is making

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I have this in my dialplan... [general] static=yes writeprotect=no clearglobalvars=no [start] exten => 5000,1,Answer exten => 5000,n,Wait(1) exten => 5000,n,NoOp(${CALLERID(num)}) exten => 5000,n,Playback(tt-monkeys) which, when I dial 5000, executes this... == Parsing '/etc/asterisk/sip_notify.conf': Found -- Executing [5000 at start:1]

Hi folks, how could I play an announcement to the calling party as soon, as the called party picked up. I would like to deploy an asterisk in an environment, where a premium rate support-number is offered to customers which do not want to pay a monthly support contract. In Germany, you are commited by law to announce the cost per minute of a premium rate number at the beginning of the call. So,

Hello, I am running Shorewall 2.0.2f, on SuSE 9.2 distro, kernel 2.6.8-24.11-default My ip addr show output is as follows: 1: lo: <LOOPBACK,UP> mtu 16436 qdisc noqueue link/loopback 00:00:00:00:00:00 brd 00:00:00:00:00:00 inet 127.0.0.1/8 brd 127.255.255.255 scope host lo inet6 ::1/128 scope host valid_lft forever preferred_lft forever 2: eth0:

I apologise for writing again about the problem with using stepAIC + multinom, but I think the reason why I had it in the first place is perhaps there may be a bug in either stepAIC or multinom. Just to repeat the problem, I have 126 variables and 99 cases. I don't know if the large number of variables could be the problem. Of couse the reason for doing a stepwise method is to reduce this

How can I objectively measure jitter in Asterisk on a SIP channel? I don't just want to turn the new 1.4 jitter buffer on. I want to measure jitter. Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070803/c6d473ce/attachment.htm

I have progressinband=yes in sip.conf, but Asterisk sends a 180-Ringing to my polycom phones and then it also sends 183-Session Progress. That doesn't seem to make sense. Shouldn't Asterisk NOT send 180-Ringing if progressinband=yes ? Doug.

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