similar to: sip interopability problem

Hi When i ran the below command on vicidial dialer: [root@vicidial2 ~]# tethereal -i eth0 -a duration:300 -w sample.cap Capturing on eth0 320167 147496 packets dropped on net i found: When i ran Acterna PVA-1000 on sample.cap it showed Max Jitter about 20 % and packet loss and echo as major cause of voice degradation. MQS was also less 2.59 where as it should be around 5.0. are packets being

I have the following in sip.conf. It was my understanding that this configuration (ie with deny/permit) would only allow connections from hosts 192.168.10.4 and 192.168.10.5. That doesn't seem to be the case. Asterisk is accepting INVITE's from other addresses. [a00090101] type=friend context=Company1 username=a00090101 ;secret=180 ;insecure=very host=dynamic mailbox=company1@vmusers

Hi, Got problems with incoming SIP calls. Scenario: Server1: 3cx or any other server Server2: Asterisk 16.2.1 . PJPROJECT 2.8 Server2 registers on Server1 with SIP ID 1121. Registration is OK. Server2 outgoing calls are OK. INVITE, unauthorized, INVITE with password, OK, RINGING,... Troubles with incoming calls / incoming INVITE's . I can not identify endpoint by IP, I have multiple

B.H. Hello, all I'm using Asterisk 11.7, connected to PSTN using SIP trunk. I'm looking for a way to get data from INVITE's SDP. Specifically, i would like to get a value of o= for incoming call from PSTN because it contains data about the operator that the call originates from. I have googled for a solution and found this patch:

Anyone know if a way to have voicemail files stored as mp3's? Thanks, Doug.

Seeking recommendations on handsets for use with Asterisk. I've been looking at the Aastra 480i CT because of its cordless handset and also the new Linksys SPA-942. Anyone using either one of these with comments on them? Any other thoughts on good reasonably priced handsets? This is for just a couple of people who work from home offices and will be connecting to an Asterisk server hosted

Hi, I use Sarge's dovecot for a pop3 mail collection server. IMAP is not used at all. It works well, but I would like to know why one of the accounts also keeps a copy of all e-mail copies in the cur/ folder? Thanks a lot, Wally -- Wally Winchester wally_winchester at fastmail.fm -- http://www.fastmail.fm - The way an email service should be

Hi, I'd like to use the convenience of apt packaging, but debian sarge's default asterisk is something like 1.0.7. Are there any apt repositories which provide newer versions of the software? Thanks! -- Jean-Michel Hiver - http://ykoz.net/ D?couvrez la R?union des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

I am trying to create a %100 g729 (with no transcoding) system (using a Soekris, of course). I am running AstLinux with the native sounds, g729 is the only codec allowed, %100 SIP (g729 only allow=) - I think I am covering all of my bases. I have only "format=g729" in voicemail.conf. On an incoming call to a mailbox, everything goes well until recording the message. When the

All, I've found problems in my setup with the latest couple of revisions (1.135/1.136) of asterisk/channels/chan_sip.c In my setup I have a RH9 asterisk server, AS5300 (single E1 to PSTN) and a dozen 7940's, everything is in the same VLAN and only running SIP. Outbound calls work fine: 7940 -SIP-> Asterisk -SIP-> AS5300 But inbound calls fail, I see the initial INVITE from the

Hello, The current dependancy on hotplug|udev (>= 0.59) seems bad. As far as I remember, last week, when I tried to install Ralph packages, I got an error from hotplug/udev and I had to use udev from backports.org Anyone know which version of udev/hotplug we will have to depends on ? Maybe we could simply rely on a recent udev dependancy, because afaik udev now provides hotplug. -- Julien

Dear Friends, We have implemented "Asterisk" in our organization. There are 150 members in our organization. At present all are using softphones. Now, I want to buy hardphones for our staff. Can anybody suggest me that what is the best hardphone for Asterisk with low-cost? Thank you. Regards, Chandra. --------------------------------- Ring'em or ping'em. Make PC-to-phone

I've being testing a couple of GrandStream ATA 286 which with no reason start responding 486 Busy to all new incoming INVITES. They are connected to an Asterisk installation as SIP client. Running ethereal between them, I could notice that, for some reason unknown for me at this time, Asterisk sends some stranges INVITE's AFTER the communication has been established and acknowledged

<gloat>Well, that''s me booked in for RailsConf in september :0)</gloat> Who else is going? Steve

Hello, Is there a common way of stripping html tags from a string? Right now I''m just calling gsub!(/<.*?>/, ''''), but with a background in PHP and always having used its strip_tags() method, I wonder if the Rails community has standardized this fairly common task with something a bit less simpleminded than my quick fix. Thanks! Zack -------------- next part

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Hi, I was wondering if anybody knew what ''<some type> expected, got Fixnum'' means? My code looks like the following: begin anevent = Event.new anevent.title = params["event"]["title"] anevent = session[:user_id] com = Community.find(params[:id]) com.events << anevent rescue Exception => exc ... end But I get an

I''m having tons of problems with fcgi on apache2. The app didn''t crash over the weekend when it wasn''t used. I ran it this morning and it was fine. It''s an inhouse app so it wasn''t used for 3 days over the weekend. It did however crash today. Does this sound like a bad fcgi setup or just a very buggy fcgi/Apache combo? Charlie Bowman

Hello, I'm having a look at section 13.1 from SIP Connect v2 doc (see [1]). It refers to RFC6442 which gives the following example (sorry for its length): INVITE sips:bob at biloxi.example.com SIP/2.0 Via: SIPS/2.0/TLS pc33.atlanta.example.com;branch=z9hG4bK74bf9 Max-Forwards: 70 To: Bob <sips:bob at biloxi.example.com> From: Alice <sips:alice at

Hi there, is there any way to get the imap and pop client to use separate keys? My Thunderbird client complains when I am popping from popss.domain.com that the certificate is owned by imapss.domain.com Any clues here? Cheers, Noah