similar to: Talking to other SIP hosts, wrong IP

I'm confused a bit about the following and was hoping to get some answers on this group - What is exactly implied when we say asterisk can connect to a PSTN. Does it mean connecting to the PSTN via PRI/T1/E1? If yes, then I assume asterisk does not need to do any SS7 signaling and all it does (playing the role of a PBX) is to connect to a Class 5 Switch at the CO. Is this a correct statement?

Hi, Is there is some form of VAD on * for SIP channels, cause I have a problem with MOH. I made an extension which simply plays MOH, when I dial that extension with my ATA188 MOH sounds choppy if I talk on the phone the MOH keeps playing. I saw the sip channel (show channel SIP/*) and I see no packets going in/out when I talk then packets shows going in/out. I don?t have this kind of problem

Hi, For MGCP users. Is there any success stories with any MGCP IAD vendor. I?m trying to find an IAD which works with Asterisk. I?ve tried the Cisco IAD 2430 without success; but SIP on this IAD works but it?s limited (no authentication, no notify messages, etc) and with higher density IAD (16 or more ports) it?s nice to control using MGCP. Any information will be apreciated ! Thanks. --

has anyone seen this? -- Starting Skinny session from 192.168.13.102 -- Starting Skinny session from 192.168.13.102 triton*CLI> Oct 15 13:44:05 WARNING[213019]: File chan_skinny.c, Line 2243 (get_input): Skinny Client sent less data than expected. Oct 15 13:44:05 NOTICE[213019]: File chan_skinny.c, Line 2301 (skinny_session): Skinny Session returned: Success Oct 15 13:44:05

Hi, Is there anybody in this list who had experience with two TE410 cards on a server ? I know that the cards can?t share IRQs and I?m seeing to have two cards on a x335 IBM Xeon server. TIA -- Juanjo sin .sig

Hello, I have finally received the details from Polycom to get into the backend configuration of their SoundPoint IP 600 SIP VOIP phone. The phone is quite nice looking but the configs are very sparse, not even a place for a secret(password) field in their SIP registration section. If anyone else has one of these and needs the passwords to get into the back end configurations, just send me an

I repost a message I put a week ago: I have a Cisco IAD 2431 which has MGCP protocol; I cannot make it to work againts Asterisk; at least there is some MGCP conversation between them but when I offhook a phone attached to IAD I get no tone at all. As anybody managed to get working Asterisk against an MGCP Cisco gateway ? Which MGCP version should I use ? Also I recently

I''m adding some code to a project using restful_authentication and acts_as_statemachine In the create method for a user there is the following code generated by the restful_authentication template @user.regster! if user.valid? And this will save and the user record if it is valid. I understand that the statemachine will then change the state from passive to pending. It will guard

Owias, The 537 is, for the most part, a drop-in replacement for the Digium card. Please search the archive and the Wiki, as there have recently been several discussions about this exact subject. To my knowledge that is about the only modem that works, but keep in mind it is _not_ supported. That being said, follow the instructions on Digium's site for installing the X100P and you should be

I install CentOS 3.4 but don't found the databases mysql and postgres where are ? Thanks

I wrapped the LDA command in a script. I can see that Postfix passes "@@mydomain.tld" as the -d argument, without quotes. I then adapted the script to specifically replace this address with "@"@mydomain.tld, but this results in the following error message by Dovecot: ??? auth: Info: userdb(?): Username character disallowed by auth_username_chars: 0x22 (username:

I upgraded to Dovecot 2.3.0.1 as advised, but it still seems broken. In the Postfix log I now see: Mar? 6 13:49:03 myhost amavis[7165]: (07165-10) K00VtLRHdrYw FWD from <admin at mydomain.tld> -> <"@"@mydomain.tld>, BODY=7BIT 250 2.0.0 from MTA(smtp:[127.0.0.1]:10025): 250 2.0.0 Ok: queued as B8CA22DA1B37 Mar? 6 13:49:03 myhost amavis[7165]: (07165-10) Passed CLEAN

Hi list, I've a strange issue with Windows 7 (also occurs on 8.1) when accessing shares on domain controllers. If I use IP address or in-domain FQDN (server.domain.name), all is right. If I use another DNS entry pointing to the same IP, share access fails with following message (translated from french) : \\somehost.somsuffix\someshare is not accessible. […] Invalid parameter Issue

Hi, I?m trying to get app_festival to work. I got the source from the Debian woody package of festival-1.4.2 and applied the patch that came with * sources it applied fine; then I made the debian package and installed it. I have this on extensions.conf: exten => 6700,1,Festival(Hi there how are you doing ?) When I dial 6700 I hear nothing and then * hangups: -- Executing

The packaging/lsb/rsync.spec file is broken as shipped: It has a "Sept" month (rpmbuild here takes only 3-letter month names), and RH gzips the manpages, so the %files list can't find them. I also added doc/README-SGML and doc/rsync.sgml to the %doc files. Patch follows. Thanks for all the good work! --- rsync-2.5.6/packaging/lsb/rsync.spec.orig 2003-01-28 06:28:35.000000000 +0100

Op 7-3-2018 om 15:40 schreef Stephan Bosch: > > > Op 7-3-2018 om 13:46 schreef Philipp Berger: >> I wrapped the LDA command in a script. I can see that Postfix passes >> "@@mydomain.tld" as the -d argument, without quotes. >> I then adapted the script to specifically replace this address with >> "@"@mydomain.tld, but this results in the

Some of you might already know that we are releasing a new phone, snom 360. To make the phone well-known and stable, we have made a soft phone version out of it and offer it for trial or private use for free (for more details, see the license conditions). There are only few limitations to the phone. First of all, the audio subsystem will work only work with an acceptable quality if you are using

Hi, I'm using the new SipAddHeader application on Asterisk 1.2.1, here's a snip of my extensions: exten => _9XXXXXXX,1,SipAddHeader(P-Asserted-Identity: <sip:${CALLERIDNUM} exten => _9XXXXXXX,2,SipAddHeader(P-Asserted-Identity: tel:${CALLERIDNUM}) exten => _9XXXXXXX,3,Dial(SIP/${EXTEN}@${SIPTRUNK},,tT) exten => _9XXXXXXX,4,Congestion The problems is that Asterisk

I have a cisco 7940 with the following sip.conf config: [Desk1.1] type=friend secret=****** defaultip=192.168.1.14 insecure=no mailbox=102 callerid="Desk1.1" qualify=500 canreinvite=no context=extensions host=dynamic group=2 I do not get message waiting indicator (mwi) on this phone. Is the another .conf file invilved in configuring this function other than the mailbox=xxx in the

I don't know if this is a problem with my cisco sip IP Phones or asterisk but I thought I would post here in case someone else has experienced this issue. When I make a call from my SIP cisco IP Phone to some remote IVRs I never get the rest of my soft keys, only the "End Call" soft key, and also DTMF doesn't work... its like the phone is acting like the remote end hasn't