search for: sipport

Hi, I have configured asterisk now with ENUM lookups which are working really perfect. Now I want to play a small soundfile before dial the number to inform the caller which protocl is used (SIP, IAX2 or ISDN). How can I do this? With Playback it doesn't seems to work: [iax2-sipport-out] ; with leading 3 using IAX-sipport exten => s,1,NoOp(Dialing ${DIALSTR} with iax2-sipport-out) exten => s,2,Answer exten => s,3,Playback(forwarded-iax) exten => s,4,Dial(IAX2/portunity-out/${DIALSTR},,trRg) exten => s,5,return Best regards, Matthias -- "Programming tod...

...;deviceProtocol>SIP</deviceProtocol> <devicePool> <callManagerGroup> <members> <member priority="0"> <callManager> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <sipPort>5060</sipPort> <securedSipPort>5061</securedSipPort> </ports> <processNodeName>SIPSERVER</processNodeName> </callManager> </member> </members> </callManagerGroup> </devicePool&...

...MAT "3" SET DIALPLAN "[234]xxx|55xxxx" SET DIALWAIT "3" SET MUSICSRVR "" SET MWISRVR "" SET PHNNUMOFSA "3" SET REGISTERWAIT 120 SET SIPDOMAIN "sip.mycompany.com" SET SIPPROXYSRVR "204.140.111.219" SET SIPPORT "5070" (this is not a typo) SET SIPREGISTRAR "204.140.111.219" SET SP_DIRSRVR 10.1.1.1 SET SP_DIRSRVRPORT 389 SET SP_DIRTOPDN ou=People,o=avaya.com IF $MODEL4 SEQ 4602 goto SETTINGS4602 IF $MODEL4 SEQ 4610 goto SETTINGS4610 IF $MODEL4 SEQ 4620 goto SETTINGS4620 IF $MODEL4 SEQ...

Hello. Virtio not sipported on current ovirt. Why? I want try to add this feature. Interesting what problems will appear.

Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written "Unprovisioned", and phone is not trying to register with asterisk. Please help!! MihaelaMJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Anyone written an extension that will take a 12 digit number, convert it to an IP address so that you can make a sip call to it. Chris

...and for "Out Bound Proxy:" use: 192.246.69.247:5082 This is how you should configure an ATA: # Set UseTftp to 0 # Set UID0 to your FWD account's FWD number (e.g. 16000). # Set PWD0 to your FWD account's password. # Set GkOrProxy to fwd.pulver.com # Set UseSIP to 1 # Set SipPort to 5060 # Set SIPRegInterval to 3500 # Set SipRegOn to 1 # Set outbound proxy to 192.246.69.247:5082 # Press the Apply button at the bottom of the web page, then # restart the ATA 186 by powering it off and then on again. I'm pretty shure I have to set the outbound proxy right. I just don...

...tpDefault>true</tftpDefault> <members> <member priority="0"> <callManager> <name>ccm-beta-5-1</name> <description>CallManager 5.0 Beta Pub - 5.0.1.032</description> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <sipPort>5060</sipPort> <securedSipPort>5061</securedSipPort> <mgcpPorts> <listen>2427</listen> <keepAlive>2428</keepAlive> </mgcpPorts> </ports> <processNodeName>ccm-beta-5-1</processNodeName> </callManager> </member> &l...

Hi, ? We are new to IP phone firmware upgradation (Sorry if it is a re-post of previous question(s)). ? Recently we have bought a cisco 7942G IP phone. It currently has SIP 42.9-0-2SR1S firmware loaded on it. We do not see any option to configure a SIP Proxy where we can provide SIP Server (Asterisk PC/Device)? IP address (with current firmware on it) to register it with Asterisk. ? Do we need to

Anyone have the 7970 xml config for sip yet? Aaron

...</dateTimeSetting> <callManagerGroup> <members> <member priority="0"> <callManager> <ports> <ethernetPhonePort>2000</ethernetPhonePort> <sipPort>5060</sipPort> <securedSipPort>5061</securedSipPort> </ports> <processNodeName>10.0.0.9</processNodeName> </callManager> </member> </members> &...

..._1 10.0.0.13 # may not be necessary if use DHCP #DnsServer_2 151.165.11.201 # may not be necessary if use DHCP RegisterExpireSec 3600 RegisterRetrySec 90RegisterExpireLimitPercent 10 FailoverRetrySec 8 SipPort 5060 SRVRecordName - #_sip._udp.unisip.com # options are ON or OFF SessionTimerSupport ON # options are ON or OFF SessionTimerRefresher ON SessionTimerMin 60 TimerInterval0 300 TimerInterval1 150...

Hi Everyone, I am having some issues getting my 7961 working with Trixbox. I have loaded the SIP code (8-3-3SR2) fine but when the phone boots up it goes into an unprovisioned state. A status message shows up and says ?Error Verifying Config Info?. I have read quite a bit on this topic (getting 7961?s to work with Asterisk and TB) and only came across a few postings where other people

I've gotten canreinvite=yes to work with a sip device behind NAT!! You *MUST* port forward the SIPPort to in your gateway router to your phone. This is a MUST. Okay, now on to my problem.. I have people who will be using ulaw, and I have people who will be using g729.. I want to set it up so that canreinivte will work.. I have a single cisco gateway.. Asterisks isn't handling the negotation be...

...f > <http://www.newace.com/asterisk/chan_sccp.c.diff> > chan_sccp.h.diff <http://www.newace.com/asterisk/chan_sccp.h.diff> > Associates "Messages" key on 79XX with a voicemail context. Adds > "voicemail" keyword to sccp.conf file. I'd done VM button sipport a while ago, just was a bit buggy, this patch has the same issue, it shouldn't be much work to fix it, and will probably go in soon. > Especially the behaviour of the speeddials is kinda annoying. The first > speeddial always starts immediately on line 1. The rest do nothing > witho...

...If I am asking the wrong person about >> this, please let me know who I should direct my questions to. Thanks. >> I am new to Samba and this ML, and I'm in trouble as well, but I am sure that the Mac OS up to 9.xxx does _NOT_ support Samba, OS 10 does. If you want to get Samba sipport on your Mac you will have to try a program called Dave. If you want to download a fully working evaluation copy of the software, then type this in your browser "http://download.thursby.com/dave252.smi.sit" it will work for about three weeks. I have Dave on my G4 Mac, and I am using Samba...

Right-- I've seen the Avaya document you cite below. It says "To administer DHCP option 242, make a copy of an existing option 176" but I don't have any example of option 176 or 242 to copy, and don't know what to do to /etc/dhcpd.conf to make it offer option 242. Then there's this long table of parameters to use with (presumably) option 242. I was hoping someone had a

...192.168.100.34 # can be an IP address or FQDN Registrar2Port 0 # enter a port number or 0 for default (5060) RegisterExpireSec 3600 Q_Param 50 RegisterExpireLimitPercent 10 Register403RetrySec 1200 SipPort 5060 SRVRecordName - #_sip._udp.uniden.com FailoverRetrySec 4 # For Redundant Outbound proxy server #InterDigitTimer Value in milli seconds (minimum is 1000ms = 1Second) InterDigitTimer 4000 # options are ON or OFF SessionTimerSupp...

My Cisco 7905 can't register with Asterisk (1.0.7-BRIstuffed-0.2.0-RC7k on Debian stable). It could, however, register with another installation of Asterisk and the settings on the phone (apart from the SIP proxy address) haven't changed since then. On the new Asterisk box my sip.conf contains this: [jeremy] type=friend regexten=801 allow=g729 host=dynamic secret=PASSWORD nat=yes

snat not working my local ip is aaa.aaa.aaa.aaa asterisk sitting on the internet at ip bbb.bbb.bbb.bbb my firewall''s internal ip is 192.168.0.254 i did snat: iptables -t nat -A POSTROUTING -o ppp0 -j SNAT --to aaa.aaa.aaa iptables -t nat -L -v gives: Chain POSTROUTING (policy ACCEPT 23663 packets, 2182K bytes) pkts bytes target prot opt in out source destination 33056