[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk
I have a peculiar problem. I have installed asterisk and also g729 (2 channels). I have a Cisco7940 IP phone with SIP installed (v6) and a cisco router 2650xm which has an isdn bri voice interface that connects to a legacy pbx system. Also I installed a x-lite to make some tests. I have configured everything after a lot of search and trial and error. So I have managed to make calls from the 7940 to x-lite and vice-versa and also to make calls to to legacy phones from the 7940 or the x-lite via the cisco router using its voice interface. BUT the problem is that from the legacy PBX phones I can call the x-lite but not the cisco 7940 IP Phone. I place the call and the cisco phone rings just once (and it shows for a fraction of second the caller id) and then the connections closes as if the called has hunged up. Where is the problem Can anyone help me? here are the configurations: SIP.CONF [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all allow=alaw allow=ulaw allow=gsm allow=g729 [xlite1] type=friend regexten=1239 ; When they register, create extension 1239 username=xlite1 callerid=Savas Pavlidis 1239 host=dynamic ;nat=yes ; X-Lite is behind a NAT router canreinvite=no; Typically set to NO if behind NAT [10.1.1.1] ; Cisco 2650XM router type=friend host=10.1.1.1 dtmfmode=rfc2833 disallow=all allow=alaw allow=g729 [419] ; 7940G Cisco IP Phone type=friend username=419 host=dynamic canreinvite=yes dtmfmode=inband disallow=all allow=g729 EXTENSIONS.CONF (PART OF IT) ; The numbers 3XX belong to the traditional ; PBX telephones. ; exten = _3XX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _3XX,n,Congestion ; ; ; exten = 419,1,Dial(SIP/419) exten = 420,1,Dial(SIP/xlite1) exten = 420,2,Congestion ; as you may understand 419 is the cisco ip phone ; and extension 420 is the softp phone x-lite ; on the pc. CISCO ROUTER CONFIGURATION (PART OF IT) dial-peer voice 1 pots destination-pattern 3.. direct-inward-dial port 1/0/0 forward-digits all ! dial-peer voice 2 pots destination-pattern 3.. direct-inward-dial port 1/0/1 forward-digits all ! ! dial-peer voice 100 voip destination-pattern 9.. session target ipv4:100.0.0.1 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 101 voip destination-pattern 8.. session target ipv4:100.0.0.1 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 103 voip destination-pattern 1.. session target ipv4:200.200.201.2 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 200 voip destination-pattern 40. session target ipv4:100.0.0.191 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 201 voip destination-pattern 5.. session target ipv4:100.0.0.191 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 202 voip destination-pattern 42. session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! dial-peer voice 205 voip destination-pattern 41. session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:10.1.1.250:5060 ! This is the SIP transaction Sip read: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 10.1.1.1:5060 From: sip:[EMAIL PROTECTED];tag=8BF5F286-8F5 To: sip:[EMAIL PROTECTED] Date: Mon, 01 Nov 2004 08:12:53 GMT Call-ID: [EMAIL PROTECTED] Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 2846660713-722735577-3128754082-2282147162 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER CSeq: 101 INVITE Max-Forwards: 6 Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=no;privacy=off Timestamp: 1099296773 Contact: sip:[EMAIL PROTECTED]:5060 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 235 v=0 o=CiscoSystemsSIP-GW-UserAgent 1213 9211 IN IP4 10.1.1.1 s=SIP Call c=IN IP4 10.1.1.1 t=0 0 m=audio 18644 RTP/AVP 8 101 c=IN IP4 10.1.1.1 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 20 headers, 11 lines Using latest request as basis request Sending to 10.1.1.1 : 5060 (non-NAT) Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 10.1.1.1:18644 Found description format PCMA Found description format telephone-event Capabilities: us - 0x10e(GSM|ULAW|ALAW|G729A), peer - audio=0x8(ALAW)/video=0x0(EMPTY), combined - 0x8(ALAW) Non-codec
[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk
Propably this is a cisco router issue. I discovered that if a put play line in the extensions.conf so that it can play something before the call is done, even for one second, the call is working normally. I also played with other variables in the sip.conf but have not succeeded except with the play line in extensions.conf exten = 419,1,Playback(pbx-transfer) exten = 419,2,Dial(SIP/419) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk
We need a little more info such as type of PBX, how it connected to *, Router type, IOS version, Router config, and at least your extensions.conf. I have setup several * using different Cisco routers and PBX's and not had any problems. I have a lab where I can setup your config with *. Thnaks. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Pavlidis Savas Sent: Monday, November 01, 2004 6:39 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk Propably this is a cisco router issue. I discovered that if a put play line in the extensions.conf so that it can play something before the call is done, even for one second, the call is working normally. I also played with other variables in the sip.conf but have not succeeded except with the play line in extensions.conf exten = 419,1,Playback(pbx-transfer) exten = 419,2,Dial(SIP/419) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk
Thanks for your reply. The cisco router is 2650xm with the following === Cisco Internetwork Operating System Software IOS (tm) C2600 Software (C2600-IS-M), Version 12.2(15)ZJ3, EARLY DEPLOYMENT RELE ASE SOFTWARE (fc2) TAC Support: http://www.cisco.com/tac Copyright (c) 1986-2003 by cisco Systems, Inc. Compiled Fri 26-Sep-03 02:05 by eaarmas Image text-base: 0x80008098, data-base: 0x81ADBC0C ROM: System Bootstrap, Version 12.2(8r) [cmong 8r], RELEASE SOFTWARE (fc1) ROM: C2600 Software (C2600-IS-M), Version 12.2(15)ZJ3, EARLY DEPLOYMENT RELEASE SOFTWARE (fc2) Yalko-Thess uptime is 11 weeks, 3 hours, 1 minute System returned to ROM by reload System restarted at 10:52:42 UTC Mon Aug 16 2004 System image file is flash:c2600-is-mz.122-15.ZJ3.bin cisco 2650XM (MPC860P) processor (revision 0x200) with 125952K/5120K bytes of me mory. Processor board ID JAE080102SE (2906160658) M860 processor: part number 5, mask 2 Bridging software. X.25 software, Version 3.0.0. Basic Rate ISDN software, Version 1.1. 1 FastEthernet/IEEE 802.3 interface(s) 1 Serial network interface(s) 3 ISDN Basic Rate interface(s) 32K bytes of non-volatile configuration memory. 32768K bytes of processor board System flash (Read/Write) Configuration register is 0x2102 == it has a VIC 2BRI-ISDN interface (NT/TE) which connects to the PBX which is a Bosch Tenovis Integral 33EW2 EuroISDN. We use already another similar cisco at another office via a WAN link to transfer data and voice channels between the two PBX's (same). I made some changes to experiment with SIP as the hardware already exists. I somewhere red that my version of Cisco IOS cannot handle the INVITE command according to RFC's. You can see a previous mail with the same header which has the configuration and the SIP transaction. I have already got the 12.3(9) and I will flash it in the first occasion and see if it corrects. In the meantime by playing a line of a small audio like transfer, does the trick and fools the cisco and works correctly. Thanks, and if you got any clues by my configs I would greatly appreciate it. Savas Pavlidis Henry Devito wrote: We need a little more info such as type of PBX, how it connected to Router type, IOS version, Router config, and at least your extensions.conf. I have setup several * using different Cisco routers and PBX's and not had any problems. I have a lab where I can setup your config with *. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] HELP: problem making calls from legacy pbx to cisco sip phone via asterisk
I have a peculiar problem. I have installed asterisk and also g729 (2 channels). I have a Cisco7940 IP phone with SIP installed (v6) and a cisco router 2650xm which has an isdn bri voice interface that connects to a legacy pbx system. Also I installed a x-lite to make some tests. I have configured everything after a lot of search and trial and error. So I have managed to make calls from the 7940 to x-lite and vice-versa and also to make calls to to legacy phones from the 7940 or the x-lite via the cisco router using its voice interface. BUT the problem is that from the legacy PBX phones I can call the x-lite but not the cisco 7940 IP Phone. Where is the problem Can anyone help me? here are the configurations: SIP.CONF [general] context=default ; Default context for incoming calls port=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls disallow=all allow=alaw allow=ulaw allow=gsm allow=g729 [xlite1] type=friend regexten=1239 ; When they register, create extension 1239 username=xlite1 callerid=Savas Pavlidis 1239 host=dynamic ;nat=yes ; X-Lite is behind a NAT router canreinvite=no; Typically set to NO if behind NAT [10.1.1.1] ; Cisco 2650XM router type=friend host=10.1.1.1 dtmfmode=rfc2833 disallow=all allow=alaw allow=g729 [419] ; 7940G Cisco IP Phone type=friend username=419 host=dynamic canreinvite=yes dtmfmode=inband disallow=all allow=g729 EXTENSIONS.CONF (PART OF IT) ; The numbers 3XX belong to the traditional ; PBX telephones. ; exten = _3XX,1,Dial(SIP/[EMAIL PROTECTED]) exten = _3XX,n,Congestion ; ; ; exten = 419,1,Dial(SIP/419) exten = 420,1,Dial(SIP/xlite1) exten = 420,2,Congestion ; as you may understand 419 is the cisco ip phone ; and extension 420 is the softp phone x-lite ; on the pc. CISCO ROUTER CONFIGURATION (PART OF IT) dial-peer voice 1 pots destination-pattern 3.. direct-inward-dial port 1/0/0 forward-digits all ! dial-peer voice 2 pots destination-pattern 3.. direct-inward-dial port 1/0/1 forward-digits all ! ! dial-peer voice 100 voip destination-pattern 9.. session target ipv4:100.0.0.1 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 101 voip destination-pattern 8.. session target ipv4:100.0.0.1 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 103 voip destination-pattern 1.. session target ipv4:200.200.201.2 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 200 voip destination-pattern 40. session target ipv4:100.0.0.191 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 201 voip destination-pattern 5.. session target ipv4:100.0.0.191 dtmf-relay h245-signal h245-alphanumeric ip qos dscp cs5 media ! dial-peer voice 202 voip destination-pattern 42. session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! dial-peer voice 205 voip destination-pattern 41. session protocol sipv2 session target sip-server session transport udp dtmf-relay rtp-nte codec g711alaw no vad ! sip-ua retry invite 3 retry response 3 retry bye 3 retry cancel 3 timers trying 1000 sip-server ipv4:10.1.1.250:5060 ! begin:vcard fn:Savas Pavlidis n:Pavlidis;Savas email;internet:[EMAIL PROTECTED] tel;work:+30 2310 573300 tel;fax:+30 2310 752280 x-mozilla-html:FALSE version:2.1 end:vcard ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
