search for: voipswitch

I am using asterisk. I also have an access to VOIPSwitch ver 2 where I can see live calls. Many times I have seen that my calls are getting strucked and then it gets disconneected after 59 mins ( as settings are done accordingly in VOIPSwitch) What could be the reason ? -------------- next part -------------- An HTML attachment was scrubbed... URL: htt...

I am trying to use asterisk as transcoder between voipswitch 2.0 and gsm gateway. Voipswitch supports g723.1 but gsm gateway does not. Now I have g723.1 codec in my asterisk. call leg from voipswitch is using codec g723.1 and call leg from gsm gateway is using codec gsm. I am having one way audio and getting below mentioned warning. Asterisk version is 1.8.1...

Hello, I'm fairly new to asterisk and Trixbox, I'm setting up a Trixbox based email to fax gateway. At this time, I have a ZAP PRI link between the eFax server and my VoIPSwitch. The ZAP channels are configured, the B and D channels are up, and I have green link lights on either end of my cabling, but when I dial the number I have assigned to my eFax server, the call never seems to route from the VoIPSwitch to the eFax server. Everything I have *looks* right, but the calls...

...8.83.xxx:5062;branch=z9hG4bK1226703311;received=82.158.83.xxx;rport=5062 From: "800902" <sip:800902 at 130.117.xxx.xxx;user=phone>;tag=467506068 To: <sip:6615xxxxx at 130.117.xxx.xxx;user=phone>;tag=as2e12c791 Call-ID: 2117388659-5062-4 at 82.158.83.xxx CSeq: 31 INVITE Server: VoIPSwitch Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Contact: <sip:6615xxxxx at 130.117.xxx.xxx> Content-Type: application/sdp Content-Length: 235 v=0 o=root 1698171141 1698171142 IN IP4 130.117.xxx.xxx s=Asterisk PBX 1.6.1.18 c=IN IP4 130.117.x...

...z9hG4bK4ae5ea5c;rport From: "2563105" <sip:412563105 at 164.77.171.XXX>;tag=as726ac50a Call-ID: 5b7470690d6cd476346bc1113f609a4b at 164.77.171.XXX To: <sip:5642325405 at 72.55.143.XXX> Contact: <sip:72.55.143.XXX:5060;transport=udp> Proxy-Authenticate: DIGEST realm="VoipSwitch", nonce="118490324119231120007472128429" Content-Length: 0 <-------------> --- (9 headers 0 lines) --- Transmitting (no NAT) to 72.55.143.XXX:5060: ACK sip:5642325405 at 72.55.143.XXX SIP/2.0 Via: SIP/2.0/UDP 164.77.171.XXX:5060;branch=z9hG4bK4ae5ea5c;rport From: "25631...

...ag=as3c6dfee5 To: <sip:5756646022 at sip.net-voz.com> Contact: <sip:5515816168 at 192.168.8.3> Call-ID: 5fc995c93d10f2a73186133377cafc88 at sip.net-voz.com CSeq: 103 INVITE User-Agent: Asterisk PBX Max-Forwards: 70 Proxy-Authorization: Digest username="5515816168", realm="VoipSwitch", algorithm=MD5, uri="sip:5756646022 at sip.net-voz.com", nonce="120404195526111105702055508208", response="cf9d8946f05b4c32a4b60aaaedd60dc8", opaque="" Date: Tue, 26 Feb 2008 16:09:09 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIF...

...switch and even though we are receiving ring from the carrier they hear no ring. We have even put a fake-ring(with Rr) back at their request and they are unable to get this ring either. The first time it happened was with a customer running a Cisco switch, now more recently we have a customer with VoipSwitch that gets no ring. Our other customers receive the ring from the carrier fine. Has anyone experienced this before and if so how did you solve it? Regards, Igor Hernandez Escape Communications.

IVR Number :17275691533 When I try it from xlite configuring my provider directly, it works perfectly. When I try to dial out from dialer , it doesnt work. [sip8] type=peer username=user fromuser=user authuser=user secret=password host=8.14.146.111 nat=no canreinvite=yes insecure=very disallow=all allow=g729 allow=ulaw context=default dtmfmode=rfc2833 What cld be the reason ? --------------

...uot; <sip:fiddialer at 59.xxx.xx.xx>;tag=as79fae976 Call-ID: 1525f0ef1e787bed51ed1ef119adb1fa at 59.xxx.xx.xx To: <sip:16785588539 at 8.14.xxx.xxx>;tag=1902000923098720982816221 Contact: <sip:8.14.xxx.xxx:5060;transport=udp> Content-Type: application/sdp Content-Length: 225 v=0 o=VoipSwitch 7220 7220 IN IP4 8.14.xxx.xxx s=VoipSIP i=Audio Session c=IN IP4 8.14.xxx.xxx t=0 0 m=audio 6220 RTP/AVP 18 101 a=rtpmap:18 G729/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (9 headers 11 lines) --- Found RTP audio format 18 Found RTP audio format 101 Peer audio RTP is a...