[Asterisk-Users] Outgoing SIP Failover
I am trying to write a outgoing Macro which has some sort of failover for failing SIP connections. For example... Try Outgoing SIP Provider 1 - No Route to Destination Try Outgoing SIP Provider 2 - Congested Try Outgoing SIP Provider 3 - Success and connect.. Everything I try doesnt work. Even if you can just point me to a good website where I can get this information.. Kind Regards, Steven Ducat. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk to a Huawei softX3000
Also have the same trouble connecting Asterisk with the Huawei softX3000 softswitch via SIP. Anyone have any experience with the Huawei switch. I am only in control of the Asterisk server, Huawei softX3000 is controlled by another company. I have included the ouput from the SJPHONE which can regsiter successfully and the Asterisk server which gets faced with 401 Unauthorized as below: SJ PHONE 2006-03-30 11:39:44.908 UDP LOCAL-219.134.98.8:5060 REGISTER sip:219.134.98.8 SIP/2.0 Via: SIP/2.0/UDP 10.27.27.18;rport;branch=z9hG4bK0a1b1b120010442bc3806986041c Content-Length: 0 Contact: sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 12 REGISTER From: sip:[EMAIL PROTECTED];tag=118201716914 Max-Forwards: 70 To: sip:[EMAIL PROTECTED] User-Agent: SJphone/1.60.289a (SJ Labs) Authorization: Digest username=2210,realm=huawei,nonce=387a3370c2354a042f4e519aac465739,uri=sip:219.134.98.8,response=023d39d9ec9417d4e1f285df758c8e27,algorithm=MD5 12:39:45 DEBUG 2006-03-30 11:39:45.033 UDP 219.134.98.8:5060-LOCAL SIP/2.0 200 OK From: sip:[EMAIL PROTECTED];tag=118201716914 To: sip:[EMAIL PROTECTED];tag=9b4ff129 CSeq: 12 REGISTER Call-ID: [EMAIL PROTECTED] Via: SIP/2.0/UDP 10.27.27.18;branch=z9hG4bK0a1b1b120010442bc3806986041c;rport=5060 Contact: sip:[EMAIL PROTECTED]:5060;user=phone;expires=3600 Content-Length: 0 ASTERISK --- (10 headers 0 lines)--- Responding to challenge, registration to domain/host name 219.134.98.8 REGISTER 13 headers, 0 lines Reliably Transmitting (no NAT) to 219.134.98.8:5060: REGISTER sip:219.134.98.8 SIP/2.0 Via: SIP/2.0/UDP 212.241.193.114:5060;branch=z9hG4bK77045b56;rport From: sip:[EMAIL PROTECTED];tag=as0519b6aa To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 105 REGISTER User-Agent: Asterisk PBX Max-Forwards: 70 Authorization: Digest username=2210, realm=huawei, algorithm=MD5, uri=sip:219.134.98.8, nonce=, response=7e0c56b053a18a956e6d26f21c934cb7, opaque= Expires: 600 Contact: sip:[EMAIL PROTECTED] Event: registration Content-Length: 0 --- Retransmitting #2 (no NAT) to 219.134.98.8:5060: OPTIONS sip:219.134.98.8 SIP/2.0 Via: SIP/2.0/UDP 212.241.193.114:5060;branch=z9hG4bK41171f7a;rport From: asterisk sip:[EMAIL PROTECTED];tag=as1613e3ad To: sip:219.134.98.8 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Thu, 30 Mar 2006 11:40:41 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Content-Length: 0 --- (0 headers 0 lines) Nat keepalive --- lvps212-241-193-114*CLI -- SIP read from 219.134.98.8:5060: SIP/2.0 401 Unauthorized From: sip:[EMAIL PROTECTED];tag=as0519b6aa To: sip:[EMAIL PROTECTED];tag=b45efef2 CSeq: 105 REGISTER Call-ID: [EMAIL PROTECTED] Via: SIP/2.0/UDP 212.241.193.114:5060;branch=z9hG4bK77045b56;rport=5060 WWW-Authenticate: Digest realm=huawei, nonce=e5dc81651624597ba53a08df5042eaca,domain=sip:huawei.com, stale=false,algorithm=MD5 Content-Length: 0 Any help is appreciated. Kind Regards, Steven Ducat. On 3/4/06, Glen Browley [EMAIL PROTECTED] wrote: Greetings, I'm having a job getting asterisk to register with a Huawei softX3000 softswitch via SIP. I keep getting 401 Unauthorized. Funny thing is I can successfully register SJPhone, a PA1688 IP Phone as well as a WiFi Phone against the switch without *any* problems. I think it's got to be something as simple as perhaps the register string which is currently usernamepassword@ip address although I've tried a number of variations without success. Here's a snipit from sip.conf allow=ulaw auth=md5 disallow=all dtmf=inband host=xxx.xxx.xxx.xxx insecure=very secret=xxx type=peer username=xxx Has anyone been able to register Asterisk against this Huawei switch? Normally I'd just muddle though it but I've spent the day working on this with NO success. I should also mention I've done ethereal dumps of devices that successfully register and I can't spot any differences. Thanks! Glen ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] CHINA DID
CHINA DID I am once again in search of China DID's. Either Shanghai (021) or Guangzhou (020). Please advise if you can supply. Steven Ducat. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] China DID Wanted
Looking for bulk DID's for the following location's in China (+86): Shanghai (021) Guangzhou (020) Shenzen (755) Also looking for bulk DID's in Hong Kong (+852). Thanks Steven Ducat. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Virtuozzo - G729
I am trying to install G729 licence on my Virtuozzo server running asterisk but I keep getting an error as it has no eth0. I get the following error when running register: [EMAIL PROTECTED] root]# /root/register G729- Digium Product Registration Copyright (C) 2004, Digium, Inc. Analyzing key 'G729-' Connecting to Digium License Server (216.207.245.3:5646)...OK Awaiting Response...OK Requesting status for 'G729-'...OK Key-ID: G729- Product: Digium-G729 Channels: 2 Demo: No Host-ID: f0:c3:f5:29:5e:ce:XX:2d:a2:6f:98:XX:6a:41:06:XX:50:f4:73:cb Unable to determine hostid. You must have at least one ethernet card [EMAIL PROTECTED] root]# Is there any way I can get the virtuozzo server to impersonate eth0. I tried the following: ln -s /etc/sysconfig/network-scripts/ifcfg-venet0 /etc/sysconfig/network-scripts/ifcfg-eth0 and restarted but I think I am off track as it had no effect. Any help would be greatly appreciated. Thanks Steve Ducat. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Cisco AS5300 -- [SIP] -- Asterisk - NO AUDIO
OK, here goes my next problem. I have puchased a DID which I can connect to via SIP I have been given the following details: Username: uka1xx Password: 1000xx Server: br.net:5160 My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT) The other end is a Cisco AS5300 (NO NAT) I can register with the Cisco with no problem. When I dial the DID it sends the call to my asterisk server and my asterisk server sends back the dial tone, no problem. The problem is when I pick up the phone, no audio. If I change the dial plan to do a Playback instead of Dial an extension I can see in the console it answers the call and starts to play the Playback but no audio. I can connect direclty to the Cisco AS5300 with sjphone or a budgetone 102 with no problem and get dial tone and full audio both ways but when I use the asterisk no audio. I have tried every codec possible. I have installed g729, g723 with no luck. I have tested both these codecs by forcing my budgetone to use with no problem so I know the codecs work. So the problem is when I ask asterisk to register to the Cisco AS5300 as a SIP Client it does everything right except pass the audio. There is no firewall configured. I know the Cisco SIP Server works because it works with the softphone SJPHONE and directly with the Budgetone 102. I have reinstalled Asterisk so many times. I have reinstalled g729 g723 so many times. The SIP debug output is pasted below. I have been struggling with this for weeks with no luck. Any help would be appreciated. Steven Ducat. * -- SIP read from 203.88.192.42:5160: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on Via: SIP/2.0/UDP 203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0 Via: SIP/2.0/UDP 211.147.240.237:5060;rport=57786 From: sip:[EMAIL PROTECTED];tag=1CA65AC-9C8 To: sip:[EMAIL PROTECTED] Date: Thu, 29 Sep 2005 20:14:40 GMT Call-ID: [EMAIL PROTECTED] Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 2153363387-811340250-2169109749-53752559 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 5 Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Timestamp: 1128024880 Contact: sip:[EMAIL PROTECTED]:57786 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 432 P-hint: Proxied P-hint: usrloc applied v=0 o=CiscoSystemsSIP-GW-UserAgent 5786 3481 IN IP4 211.147.240.237 s=SIP Call c=IN IP4 211.147.240.237 t=0 0 m=audio 37708 RTP/AVP 18 4 3 8 0 110 c=IN IP4 203.88.192.42 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:110 X-NSE/8000 a=fmtp:110 192-194 a=direction:passive a=direction:active a=nortpproxy:yes --- (24 headers 19 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 203.88.192.42 : 5160 (non-NAT) Found no matching peer or user for '203.88.192.42:5160' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 110 Peer audio RTP is at port 211.147.240.237:37708 Found description format G729 Found description format G723 Found description format GSM Found description format PCMA Found description format PCMU Found description format X-NSE Capabilities: us - 0x100 (g729), peer - audio=0x30f (g723|gsm|ulaw|alaw|g729|speex)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 84104214 in default (domain 70.84.200.204) list_route: hop: sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on Transmitting (no NAT) to 203.88.192.42:5160: SIP/2.0 100 Trying Via: SIP/2.0/UDP 203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0;received=203.88.192.42 Via: SIP/2.0/UDP 211.147.240.237:5060;rport=57786 From: sip:[EMAIL PROTECTED];tag=1CA65AC-9C8 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Executing Dial(SIP/211.147.240.237-b7116c10, Local/2001/n) in new stack -- Executing Macro(Local/[EMAIL PROTECTED],2, oneline|SIP/stevenducat) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/stevenducat|20) in new stack -- Called 2001/n We're at 70.84.200.204 port 14922 Answering/Requesting with root capability 0x100 (g729) 12 headers, 8 lines Reliably Transmitting (NAT) to 83.146.11.93:60073: INVITE sip:[EMAIL PROTECTED]:18234 SIP/2.0 Via: SIP/2.0/UDP 70.84.200.204:5060;branch=z9hG4bK502b5287;rport From: 0017911 sip:[EMAIL PROTECTED];tag=as2c8caf36 To: sip:[EMAIL PROTECTED]:18234 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL
[Asterisk-Users] Cisco AS5300 -- [SIP] -- Asterisk - NO AUDIO
OK, here goes my next problem. I have puchased a DID which I can connect to via SIP I have been given the following details: Username: uka1xx Password: 1000xx Server: br.net:5160 My equipment is Asterisk CVS HEAD on Red Hat EL 3.0 (NO NAT) The other end is a Cisco AS5300 (NO NAT) I can register with the Cisco with no problem. When I dial the DID it sends the call to my asterisk server and my asterisk server sends back the dial tone, no problem. The problem is when I pick up the phone, no audio. If I change the dial plan to do a Playback instead of Dial an extension I can see in the console it answers the call and starts to play the Playback but no audio. I can connect direclty to the Cisco AS5300 with sjphone or a budgetone 102 with no problem and get dial tone and full audio both ways but when I use the asterisk no audio. I have tried every codec possible. I have installed g729, g723 with no luck. I have tested both these codecs by forcing my budgetone to use with no problem so I know the codecs work. So the problem is when I ask asterisk to register to the Cisco AS5300 as a SIP Client it does everything right except pass the audio. There is no firewall configured. I know the Cisco SIP Server works because it works with the softphone SJPHONE and directly with the Budgetone 102. I have reinstalled Asterisk so many times. I have reinstalled g729 g723 so many times. The SIP debug output is pasted below. I have been struggling with this for weeks with no luck. Any help would be appreciated. Steven Ducat. * -- SIP read from 203.88.192.42:5160: INVITE sip:[EMAIL PROTECTED] SIP/2.0 Record-Route: sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on Via: SIP/2.0/UDP 203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0 Via: SIP/2.0/UDP 211.147.240.237:5060;rport=57786 From: sip:[EMAIL PROTECTED];tag=1CA65AC-9C8 To: sip:[EMAIL PROTECTED] Date: Thu, 29 Sep 2005 20:14:40 GMT Call-ID: [EMAIL PROTECTED] Supported: timer,100rel Min-SE: 1800 Cisco-Guid: 2153363387-811340250-2169109749-53752559 User-Agent: Cisco-SIPGateway/IOS-12.x Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO CSeq: 101 INVITE Max-Forwards: 5 Remote-Party-ID: sip:[EMAIL PROTECTED];party=calling;screen=yes;privacy=off Timestamp: 1128024880 Contact: sip:[EMAIL PROTECTED]:57786 Expires: 180 Allow-Events: telephone-event Content-Type: application/sdp Content-Length: 432 P-hint: Proxied P-hint: usrloc applied v=0 o=CiscoSystemsSIP-GW-UserAgent 5786 3481 IN IP4 211.147.240.237 s=SIP Call c=IN IP4 211.147.240.237 t=0 0 m=audio 37708 RTP/AVP 18 4 3 8 0 110 c=IN IP4 203.88.192.42 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=fmtp:4 annexa=no a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:110 X-NSE/8000 a=fmtp:110 192-194 a=direction:passive a=direction:active a=nortpproxy:yes --- (24 headers 19 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Sending to 203.88.192.42 : 5160 (non-NAT) Found no matching peer or user for '203.88.192.42:5160' Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 0 Found RTP audio format 110 Peer audio RTP is at port 211.147.240.237:37708 Found description format G729 Found description format G723 Found description format GSM Found description format PCMA Found description format PCMU Found description format X-NSE Capabilities: us - 0x100 (g729), peer - audio=0x30f (g723|gsm|ulaw|alaw|g729|speex)/video=0x0 (nothing), combined - 0x100 (g729) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing) Looking for 84104214 in default (domain 70.84.200.204) list_route: hop: sip:203.88.192.42:5160;ftag=1CA65AC-9C8;lr=on Transmitting (no NAT) to 203.88.192.42:5160: SIP/2.0 100 Trying Via: SIP/2.0/UDP 203.88.192.42:5160;branch=z9hG4bK5aeb.8f7e6027.0;received=203.88.192.42 Via: SIP/2.0/UDP 211.147.240.237:5060;rport=57786 From: sip:[EMAIL PROTECTED];tag=1CA65AC-9C8 To: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 101 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- -- Executing Dial(SIP/211.147.240.237-b7116c10, Local/2001/n) in new stack -- Executing Macro(Local/[EMAIL PROTECTED],2, oneline|SIP/stevenducat) in new stack -- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/stevenducat|20) in new stack -- Called 2001/n We're at 70.84.200.204 port 14922 Answering/Requesting with root capability 0x100 (g729) 12 headers, 8 lines Reliably Transmitting (NAT) to 83.146.11.93:60073: INVITE sip:[EMAIL PROTECTED]:18234 SIP/2.0 Via: SIP/2.0/UDP 70.84.200.204:5060;branch=z9hG4bK502b5287;rport From: 0017911 sip:[EMAIL PROTECTED];tag=as2c8caf36 To: sip:[EMAIL PROTECTED]:18234 Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL
[Asterisk-Users] Looking for China DID
I am looking for a China DID so my family in China can call me in UK. I am looking for an option where the providor can forward me the calls directly to my * box by SIP or IAX2 (fixed IP). Any help would be appreciated. Steven Ducat. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] oh323 or h323
LTenorio, I am not sure what you mean between Terminal and Gateway. The voip providor in China sell the H323 service as a package where you get a h323 compatible handset and a landline number, the phone comes preconfigured to connect to their gatekeeper to make and receive calls. So what I want to do is get * to pretend its the handset, register with the gatekeeper so when anyone calls the landline number the gatekeeper passes the call to my * box and when I want to call China I can pass the outgoing call to the h323 gatekeeper. I have bought 2 numbers so I need to have * pretend its 2 different handsets to receive calls from both numbers. The voip provider has given me the following details.. Protocol = H323 Gatekeeper = 210.21.118.xxx H323ID = .HMA0200.10szxn-hxxx e164 = 02022xx2912 H323ID = .HMA0200.10szxn-kxxx e164 = 02022xx2913 Thanks for your help, it is much appreciated. Kind Regards, Steve Ducat. On 9/1/05, Leandro Tenorio [EMAIL PROTECTED] wrote: I'm using oh323 too without any issues, but in Steve specific configuration, depends on how his provider expect to be register as (Terminal or Gw) afaik, oh323 just could be binded as gateway, so better ask the provider. LTenorio From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mehdi chouikh Sent: Thursday, September 01, 2005 12:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] oh323 or h323 Hello Personaly i prefer oh323, i am using for one year whitout problems. and is more easier to configure. regards On 9/1/05, Steve Ducat [EMAIL PROTECTED] wrote: I have just signed up for 2 landline numbers in China. They have offered to sell me 2 h323 compatible handsets which I have declined as I want these numbers to ring into my * box. They have given me the following info (modified for security).. Protocol = H323 Gatekeeper = 210.21.118.xxx H323ID = .HMA0200.10szxn-hxxx e164 = 02022xx2912 H323ID = .HMA0200.10szxn-kxxx e164 = 02022xx2913 Really what I want is for * to act as the endpoint. So the big question, do I use oh323 or h323 or something else. I am all confused about who is the gatekeeper, who is the gateway. I just want * to register with the gatekeeper so they will pass * all the incoming calls. Which one do I use and how would I tackle the conf file to register with the gatekeeper. Any help would be appreciated. Steve Ducat. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] oh323 or h323
LTenorio, Then this is my problem. I can only register to the gatekeeper as a terminal, they do not allow me to register as a gateway. Is there any other way I can get asterisk to register to the gatekeeper as a terminal. Thanks again for your help. Kind Regards, Steven Ducat. On 9/2/05, Leandro Tenorio [EMAIL PROTECTED] wrote: To be simple, you can register to any gk as Gateway or as Terminal, some gk restricts the way the users could be registered as. Oh323 advertise itself as gateways and if your provider does not support registration as GW it will not work. LTenorio -Original Message- From: Steve Ducat [mailto:[EMAIL PROTECTED] Sent: Friday, September 02, 2005 5:59 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: [EMAIL PROTECTED] Subject: Re: [Asterisk-Users] oh323 or h323 LTenorio, I am not sure what you mean between Terminal and Gateway. The voip providor in China sell the H323 service as a package where you get a h323 compatible handset and a landline number, the phone comes preconfigured to connect to their gatekeeper to make and receive calls. So what I want to do is get * to pretend its the handset, register with the gatekeeper so when anyone calls the landline number the gatekeeper passes the call to my * box and when I want to call China I can pass the outgoing call to the h323 gatekeeper. I have bought 2 numbers so I need to have * pretend its 2 different handsets to receive calls from both numbers. The voip provider has given me the following details.. Protocol = H323 Gatekeeper = 210.21.118.xxx H323ID = .HMA0200.10szxn-hxxx e164 = 02022xx2912 H323ID = .HMA0200.10szxn-kxxx e164 = 02022xx2913 Thanks for your help, it is much appreciated. Kind Regards, Steve Ducat. On 9/1/05, Leandro Tenorio [EMAIL PROTECTED] wrote: I'm using oh323 too without any issues, but in Steve specific configuration, depends on how his provider expect to be register as (Terminal or Gw) afaik, oh323 just could be binded as gateway, so better ask the provider. LTenorio From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mehdi chouikh Sent: Thursday, September 01, 2005 12:11 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] oh323 or h323 Hello Personaly i prefer oh323, i am using for one year whitout problems. and is more easier to configure. regards On 9/1/05, Steve Ducat [EMAIL PROTECTED] wrote: I have just signed up for 2 landline numbers in China. They have offered to sell me 2 h323 compatible handsets which I have declined as I want these numbers to ring into my * box. They have given me the following info (modified for security).. Protocol = H323 Gatekeeper = 210.21.118.xxx H323ID = .HMA0200.10szxn-hxxx e164 = 02022xx2912 H323ID = .HMA0200.10szxn-kxxx e164 = 02022xx2913 Really what I want is for * to act as the endpoint. So the big question, do I use oh323 or h323 or something else. I am all confused about who is the gatekeeper, who is the gateway. I just want * to register with the gatekeeper so they will pass * all the incoming calls. Which one do I use and how would I tackle the conf file to register with the gatekeeper. Any help would be appreciated. Steve Ducat. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] oh323 or h323
OK, now I am getting closer.
I am trying to get asterisk to connect to a gatekeeper as a terminal
to receive calls made to the landline number in china which is passed
along via h323.
I have now concentrated on only 1 number.
Protocol = H323
Gatekeeper = 210.21.118.xxx
H323ID = .HMA0200.10szxn-hxxx
e164 = 02022xx2912
I can now register with the gatekeeper successfully. Now when I call
the landline number which is routed through to my h323 connection I
hear a message saying the other caller has hung up and when I show
trace (Level 20) in the console I see the following appear:
2:00.031 Transactor:a091e98 h323pdu.cxx(501) H225RAS
Receiving PDU:
nonStandardMessage {
requestSeqNum = 5
nonStandardData = {
nonStandardIdentifier = h221NonStandard {
t35CountryCode = 38
t35Extension = 2
manufacturerCode = 68
}
data = 19 octets {
32 31 30 2e 32 31 2e 31 31 38 2e 32 32 30 3a 38 210.21.118.xxx:8
30 30 30 000
}
}
}
Raw PDU:
size=1c pos=1c.0 {
000 5c 00 04 40 26 02 00 44 13 32 31 30 2e 32 31 2e \ @ D 210.21.
31 31 38 2e 32 32 30 3a 38 30 30 30 118.xxx:8000
}
2:00.032 Transactor:a091e98h323trans.cxx(343) Trans
Reading PDU
Obviously my asterisk box does not respond to this. It continues to
reregister with the gatekeeper every minute or so but everytime I call
the landline number the only thing that happens is the above appears
in the trace.
I am now using the h323 channel.
Am I on the right track or have I lost the plot.
My config is (h323.conf):
[general]
port = 1720
bindaddr = 70.84.200.xxx
allow=all
gatekeeper = 210.21.118.xxx
AllowGKRouted = yes
context=default
h323id=.HMA0200.10szxn-
[.HMA0200.10szxn-]
type=h323
e164=02022xx2912
context=default
Thanks
Steve Ducat.
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[Asterisk-Users] oh323 or h323
I have just signed up for 2 landline numbers in China. They have offered to sell me 2 h323 compatible handsets which I have declined as I want these numbers to ring into my * box. They have given me the following info (modified for security).. Protocol = H323 Gatekeeper = 210.21.118.xxx H323ID = .HMA0200.10szxn-hxxx e164 = 02022xx2912 H323ID = .HMA0200.10szxn-kxxx e164 = 02022xx2913 Really what I want is for * to act as the endpoint. So the big question, do I use oh323 or h323 or something else. I am all confused about who is the gatekeeper, who is the gateway. I just want * to register with the gatekeeper so they will pass * all the incoming calls. Which one do I use and how would I tackle the conf file to register with the gatekeeper. Any help would be appreciated. Steve Ducat. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] Register Asterisk with Gatekeeper - oh323
Michael, thanks for your reply.. I am still having trouble connecting to the gatekeeper to receive any calls... Here is my setup... Asterisk 1.0.9 pwlib-Janus openh323-Janus asterisk-oh323-0.6.6 Here is my oh323.conf [general] listenAddress=70.84.200.xxx listenPort=1720 gatekeeper=210.21.118.xxx gatekeeperTTL=600 tcpStart=1 tcpEnd=2 udpStart=1 udpEnd=2 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=10 libTraceLevel=5 libTraceFile=stdout userInputMode=TONE amaFlags=default accountCode=H323 language=en [register] context=default alias=.HMA0200.10szxn- alias=22xx2570 alias=.HMA0200.10szxn- alias=22xx2913 [codecs] codec=G711A frames=2 codec=G729 frames=2 codec=GSM0610 frames=4 codec=G7231 frames=2 [22xx2912] type=friend ip=210.21.118.xxx ;port=1720 alias=.HMA0200.10szxn- h323id=.HMA0200.10szxn- e164=22xx2912 context=default ;disallow=all ;allow=ulaw dtmfmode=rfc2833 [22xx2913] type=friend ip=210.21.118.xxx ;port=1720 alias=.HMA0200.10szxn- h323id=.HMA0200.10szxn- e164=22xx2913 context=default ;disallow=all ;allow=ulaw dtmfmode=rfc2833 I can now register with the gatekeeper successfully but the gatekeeper will not pass me any calls. I am trying to connect to a AVS Gatekeeper by Auvtech. The details they have given me is (2 different numbers): H323ID = .HMA0200.10szxn- e164 = 22xx2912 H323ID = .HMA0200.10szxn- e164 = 22xx2913 Any more help would be appreciated. Kind Regards, Steven Ducat. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Register Asterisk with Gatekeeper - oh323
I have tried everything. to register with this gatekeeper to make and receive calls These are the details I received from the voip provider: protocol H.323 Gatekeeper Address - [EMAIL PROTECTED] Port - 1719 RAS - 53 Q931 - 80 h245 - 1722 RTP - 1722 Username - H323 I have 2 phone number/accounts with this gatekeeper that I need to register to. ID - HMA0200.10szxn- e.164 - 22xx2912 ID - HMA0200.10szxn- e.164 - 22xx2913 Here is my oh323.conf: [general] listenAddress=0.0.0.0 listenPort=1720 [EMAIL PROTECTED] gatekeeperTTL=600 tcpStart=1 tcpEnd=2 udpStart=1 udpEnd=2 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout userInputMode=TONE amaFlags=default accountCode=H323 language=en context=voip-h323 [register] alias=ASTERISK [codecs] codec=G711A frames=20 [22xx2912] type=friend [EMAIL PROTECTED] port=1720 alias=HMA0200.10szxn- e164=22xx2912 context=default disallow=all allow=ulaw dtmfmode=rfc2833 [22xx2913] type=friend [EMAIL PROTECTED] port=1720 alias=HMA0200.10szxn- e164=22xx2913 context=default disallow=all allow=ulaw dtmfmode=rfc2833 All I get from Asterisk is the following: Aug 29 10:00:57 WARNING[9715]: chan_oh323.c:4228 oh323_gk_check: Failed to register with gatekeeper '[EMAIL PROTECTED]'. -- Retrying gatekeeper registration. Am I on the right track or have I missed the point. I do not want Asterisk to be the gatekeeper, I simply want Asterisk to register with the gatekeeper so I can receive calls from it and then use this gatekeeper to make calls to it. Any help would be appreciated. Thanks Steve.. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] RxFax - IAX (Telappliant)
I am using Telappliant PSTN - VOIP Goegraphic number. They are passing me the call to my * box by IAX. I have configured spandsp, libtiff, etc. It will pick up the call, start to talk to the fax machine but at the same point every time it hangs on and then hangsup. Here is my iax.conf: [0207100] type=friend username=0207100 context=default disallow=all allow=ulaw Here is the output when receiving the fax: -- Accepting unauthenticated call from 217.14.132.185, requested format = 8, actual format = 4 -- Executing RxFAX(IAX2/[EMAIL PROTECTED]:4569/1, /var/spool/asterisk/incoming/2131665134.tif) in new stack Changed from phase 0 to 1 Slow carrier up Slow carrier down Start receiving document Changed from phase 1 to 4 Sending ident CSI: 40 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 20 DIS: Preferred octets: 256 Can receive fax Supported data signalling rates: V.27ter and V.29 R8x7.7lines/mm and/or 200x200pels/25.4mm OK 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Receiver's minimum scan line time: 0ms at 3.85 l/mm: T7.7 = T3.85 R8x15.4lines/mm OK Minimum scan line time for higher resolutions: T15.4 = T7.7 DIS: 80 00 ce f0 80 80 01 HDLC underflow in state 9 Changed from phase 4 to 3 Slow carrier up TSI: 43 31 37 35 32 20 30 38 35 20 31 37 31 20 34 34 20 20 20 20 20 TSI without final frame tag Remote fax gave TSI as: 44 171 580 2571 DCS: 83 00 86 f0 80 80 00 DCS with final frame tag In state 9 DCS: Can receive fax Selected data signalling rate: V.29, 9600bps 2D coding OK Scan line length: 215mm Recording length: A4 (297mm) Minimum scan line time: 0ms Minimum scan line time for higher resolutions: T15.4 = T7.7 Get at 9600 Changed from phase 3 to 5 Fast carrier up Training failed (sequence failed) Fast carrier training failed Fast carrier down Fast carrier up Coarse carrier frequency 1699.94 (64) Training error 7.908183 Training succeeded (constellation mismatch 12.376823) Fast carrier trained Fast carrier down Changed from phase 5 to 4 Start rx document - compression 2 Start rx page CFR: 84 HDLC underflow in state 5 Post trainability Changed from phase 4 to 5 Fast carrier up Coarse carrier frequency 1700.03 (64) Training error 7.017459 Training succeeded (constellation mismatch 10.231490) Fast carrier trained -- Hungup 'IAX2/[EMAIL PROTECTED]:4569/1' Any help would be appreciated. Steve ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
