search for: cirpack

Hi, I've got an * box with 4 E1 (TE405P) and a Class 4 Cirpack switch (www.cirpack.com). <IP Network>--<*>--<Cirpack>--<Public PSTN Network> ISDN interco is EuroIsdn, ports are configured ccs, hdb3, crc4. Cirpack is Network, * is Terminal/User. As I encountered some pb with Sip to Zap transcoding (* to Cirpack way poor quality, th...

Hi, has anyone connected Asterisk to Cirpack softswitch sucessfully ? Any howto or more info about needed Asterisk SW and setup ? Thanks in advance, regards, Rob.

...v1.4.14, my message log is now filling up with the following error messages: <-------------> [Dec 29 17:24:52] WARNING[10655]: chan_sip.c:6645 determine_firstline_parts: Bad request protocol Packet --- (1 headers 0 lines) --- bitis*CLI> <--- SIP read from 82.101.62.99:5060 ---> Cirpack KeepAlive Packet <-------------> Seeing as these packets are being sent by one of my service providers, I can't just turn them off. What's the best solution for this problem? Thanks, Jaap

Hi all, I've a problem in DTMF dialog between * and a Cicpack Transit Node Class 4-5. The call initiat by a mgcp phone pass by the cirpack and arrive in SIP on *, everything is ok (negociation and phone call) but when we try to use the voicemail, Asterisk don't understand DTMF. Here are some logs (SIP debug on) on a DTMF '2' receive : ************************************************************************* Sip read: I...

Hi, I've got an * box with 4 E1 (TE405P) and a Class 4 Cirpack switch (www.cirpack.com). <IP Network>--<*>-[TE405P]-<Cirpack>-<Public PSTN Network> ISDN interco is EuroIsdn, ports are configured ccs, hdb3, crc4. Cirpack is Network, * is Terminal/User. As I encountered some pb with Sip to Zap transcoding (* to Cirpack way poor qual...

Salut Guy, I have the same problem with a Cirpack (B3G carrier) What I see is that you use sip info to detect DTMF. The problem is that there is no normalisation on the content of the sip info frame for dtmf detection. First, asterisk try to detect the header "application/dtmf-relay" and you have the header "application/dtmf&quot...

Salut Guy, I have the same problem with a Cirpack (B3G carrier) What I see is that you use sip info to detect DTMF. The problem is that there is no normalisation on the content of the sip info frame for dtmf detection. First, asterisk try to detect the header "application/dtmf-relay" and you have the header "application/dtmf&quot...

I was messing about integration of a Cirpack softswitch with Asterisk and banged my head against a problem previously noted on the list. http://lists.digium.com/pipermail/asterisk-users/2003-November/026436.ht ml What is the status of this problem? Has it been fixed? I scrambled through chan_sip.c, but couldn't find ay reference to &qu...

...provider. One box runs without errors on the console, the other box keeps repeating : [Feb 11 23:40:29] WARNING[11292]: chan_sip.c:6705 determine_firstline_parts: Bad request protocol Packet When i set debug on, it seems to come from that SIP DID. <--- SIP read from 82.101.62.99:5060 ---> Cirpack KeepAlive Packet <-------------> [Feb 11 23:37:59] WARNING[11292]: chan_sip.c:6705 determine_firstline_parts: Bad request protocol Packet --- (1 headers 0 lines) --- What i don't understand is why i get this message on one box only? Ideas anyone?

...837515 INVITE From: "033426aaaaaa" <sip:033426aaaaaa at sip.myoperator.net;user=phone>;tag=04459-CI-5fa6f8a1-6f03b5b60 Max-Forwards: 30 P-Preferred-Identity: <sip:033426aaaaaa at sip.myoperator.net;user=phone> To: <sip:081169xxxx at 91.121.xxx.xxx;user=phone> User-Agent: Cirpack/v4.42s (gw_sip) Via: SIP/2.0/UDP 91.121.xxx.xxx:5060;branch=z9hG4bK-4B4B-2D82DB2 Content-Length: 481 v=0 o=cp10 130085910854 130085910854 IN IP4 10.7.1.121 s=SIP Call c=IN IP4 91.121.bbb.bbb t=0 0 m=audio 36146 RTP/AVP 18 4 0 8 125 111 101 b=AS:21 a=rtpmap:18 G729/8000/1 a=fmtp:18 annexb=no a=rtpm...

When I get a SIP call, I get the following error: *CLI> NOTICE[1133718080]: File chan_sip.c, Line 1768 (process_sdp): Content is 'multipart/mixed;boundary="unique-boundary-1"', not 'application/sdp' WARNING[1225991360]: File pbx.c, Line 1154 (pbx_extension_helper): No application '' for extension (incoming, 5147771111, 1) == Spawn extension (incoming,

Hi, I'm trying to get asterisk to work with the Cirpack Softswitch. All I need for now is that asterisk should forward all calls to the Cirpack. My sip.conf files looks like: [general]

...>;tag=25151-GA-0eaf098c-32a97dc05 Max-Forwards: 28 Record-Route: <sip:C=on-88.165.134.117.5060;t=RDKIW at 212.27.52.5:5060;lr> To: <sip:095199YYYY at 172.17.20.241;user=phone> Via: SIP/2.0/UDP 212.27.52.5:5060;branch=z9hG4bK-RDKI-00720204-55801b4f Allow: UPDATE,REFER,INFO User-Agent: Cirpack/v4.41c (gw_sip) Content-Length: 173 v=0 o=cp10 125830752022 125830752022 IN IP4 212.27.52.129 s=SIP Call c=IN IP4 212.27.52.129 t=0 0 m=audio 36480 RTP/AVP 8 b=AS:64 a=rtpmap:8 PCMA/8000/1 a=ptime:30 <-------------> --- (13 headers 9 lines) --- == Using SIP RTP CoS mark 5 Sending to 212.2...

...5060;lr;nat=yes>. From: "0161801019" <sip:0161801019@192.168.0.173>;tag=as02de1b95. To: <sip:0161888874@10.254.254.1>;tag=00-04094-52dbe3bc-6cf68a723. Call-ID: 71f7297e0e6cc0625bbae5be00f8a2cc@192.168.0.173. CSeq: 102 INVITE. Contact: <sip:212.241.48.70:5060>. server: Cirpack/v4.38f (gw_sip). Allow: UPDATE, REFER. Content-Type: application/sdp. Content-Length: 253. . v=0. o=cp10 112775383044 112775383045 IN IP4 10.166.38.109. s=SIP Call. c=IN IP4 10.254.254.1. t=0 0. m=audio 35058 RTP/AVP 18 101. b=AS:64. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 teleph...

...od: REGISTER [Mar 20 18:34:49] VERBOSE[7840] logger.c: <--- SIP read from xxx.xxx.xxx.xxx:5060 ---> SIP/2.0 423 Interval Too Brief Call-ID: 522ef0f7118f00d83d42b1082882fc10 at 192.168.0.1 CSeq: 174 REGISTER From: <sip:XXXXXXXX@ xxx.xxx.xxx.xxx>;tag=as200dbc2c Min-Expires: 600 Server: Cirpack/v4.41f (gw_sip) To: <sip:59972778@ xxx.xxx.xxx.xxx>;tag=00-08013-1313fd60-3a4260273 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;received= xxx.xxx.xxx.xxx;rport=5060;branch=z9hG4bK259bcd00 WWW-Authenticate: Digest realm="XXX.XXX",nonce="1313fbd315b1cefb52c870440e6f5455",opaque...

...com.com. Contact: <sip:217.66.118.145:5060>. Content-Type: application/sdp. CSeq: 34878212 INVITE. From: "0614740696" <sip:0614740696 at sip.lecom.com;user=phone>;tag=02975-US-0223ae6e-67d6c4495. Max-Forwards: 31. To: <sip:0170080048 at 127.0.0.1;user=phone>. User-Agent: Cirpack/v4.41c (gw_sip). Via: SIP/2.0/UDP 217.36.112.145:5060;branch=z9hG4bK-744D-33B812. Content-Length: 303. . Whereas with this one I can do it! (there is a number in the INVITE) # U 87.98.202.114:5060 -> 192.168.95.235:5060 INVITE sip:0170704626 at 192.168.95.235 SIP/2.0. Via: SIP/2.0/UDP 87.98.2...

....xs4all.nl Contact: <sip:82.101.62.99:5060> Content-Type: application/sdp CSeq: 103 INVITE From: "**********" <sip:**********@sip.xs4all.nl>;tag=as70e84199 Record-Route: <sip:82.101.62.115;lr;r2=on;ftag=as70e84199>,<sip:82.101.63.5;lr;r2=on;ftag=as70e84199> Server: Cirpack/v4.41b (gw_sip) To: <sip:0031********@sip.xs4all.nl>;tag=00-08168-044b6f36-245cd72c7 Via: SIP/2.0/UDP ***.***.***.***:5060;received=***.***.***.***;rport=5060;branch=z9hG4bK07c2ed92 Content-Length: 182 v=0 o=cp10 125193221174 125193221174 IN IP4 82.101.62.66 s=SIP Call c=IN IP4 194.109.8.2 t...

...rom: <sip:0001@somehost.somedomain.tdl>;tag=as32d2cdfe Proxy-Authenticate: Digest realm="freephonie.net",nonce="012dd3995b84e8f56ca34a7201a0c6ff",opaque="012daad2220ed2c",stale=false,algorithm=MD5 Record-Route: <sip:24.196.79.163;lr;ftag=as32d2cdfe> Server: Cirpack/v4.40 (gw_sip) To: <sip:*759608xxxxxxxx@sipbroker.com>;tag=01-08146-012dd3ab-3b2383163 Via: SIP/2.0/UDP 172.16.1.1:5060;received=86.216.233.69;rport=5060;branch=z9hG4bK76bd560d Content-Length: 0 --- (12 headers 0 lines)--- Transmitting (no NAT) to 24.196.79.163:5060: ACK sip:*759608xxxxxxxx...

...5060;lr;nat=yes>. From: "0161801019" <sip:0161801019@192.168.0.173>;tag=as02de1b95. To: <sip:0161888874@10.254.254.1>;tag=00-04094-52dbe3bc-6cf68a723. Call-ID: 71f7297e0e6cc0625bbae5be00f8a2cc@192.168.0.173. CSeq: 102 INVITE. Contact: <sip:212.241.48.70:5060>. server: Cirpack/v4.38f (gw_sip). Allow: UPDATE, REFER. Content-Type: application/sdp. Content-Length: 253. . v=0. o=cp10 112775383044 112775383045 IN IP4 10.166.38.109. s=SIP Call. c=IN IP4 10.254.254.1. t=0 0. m=audio 35058 RTP/AVP 18 101. b=AS:64. a=rtpmap:18 G729/8000/1. a=fmtp:18 annexb=no. a=rtpmap:101 teleph...

Hi folks, Following an upgrade from Debian squeeze to wheezy, and Asterisk 1.6.2.9 to 1.8.13, my server is no longer able to register a connection to a SIP account at my ISP (XS4ALL in the Netherlands). At the same time, it is still able to register a different account with another SIP provider, so it must be that they no longer have the same basic requirements. The relevant part of my