Re: [asterisk-users] Attempting native bridge of
El vie, nov 17 de 2006 a las 04:34 +0530, Vicky comentaba: Thats really strange .. if you have made canreinvite=no then it should not even attampt native bridging and should transcode codecs ..something's fishy here .. Also try to put canreinvite=no in testulaw exntension too . So why do I have audio in both ways using 2 IP Phones with 2 different codecs and getting 'Attempting native bridging' at the same time? I've always had canreinvite=no in all my extensions. This is my sip.conf: [testgsm] type=friend host=dynamic username=testgsm context=astertest canreinvite=no disallow=all allow=gsm [testulaw] type=friend host=dynamic username=testulaw context=astertest canreinvite=no disallow=all allow=ulaw [testalaw] type=friend host=dynamic username=testalaw context=astertest canreinvite=no disallow=all allow=alaw [testg723] type=friend host=dynamic username=testg723 context=astertest canreinvite=no disallow=all allow=g723 [testg729] type=friend host=dynamic username=testg729 context=astertest canreinvite=no disallow=all allow=g729 [1001] type=friend host=dynamic username=1001 context=astertest canreinvite=no disallow=all allow=g729 [1010] type=friend host=dynamic username=1010 context=astertest canreinvite=no disallow=all allow=ulaw Thanks again! ___ --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] Attempting native bridge of
g729 is not a free codec . YOu have to buy it from digium at rateof $10 per channel license . If you are just using asterisk and havent bought g729 license then asterisk will just do bridging of g729 and wont edit/transcode stream . On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote: I have the following scenario: g729gsm UAS --- * --- UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ¿Am I wrong? The UAC and UAS are registering with * properly: --- sip.conf [testgsm] type=friend host=dynamic username=testgsm context=astertest canreinvite=no disallow=all allow=gsm [testg729] type=friend host=dynamic username=testg729 context=astertest canreinvite=no disallow=all allow=g729 - - dspam*CLI sip show peers Name/username HostDyn Nat ACL Port Status testgsm/testgsm172.16.51.244D 1 Unmonitored testg729/testg729 172.16.51.244D 2 Unmonitored -- Executing Answer(SIP/testgsm-081784b0, ) in new stack -- Executing Wait(SIP/testgsm-081784b0, 1) in new stack -- Executing Dial(SIP/testgsm-081784b0, SIP/testg729) in new stack -- Called testg729 -- SIP/testg729-0817dd90 is ringing -- SIP/testg729-0817dd90 answered SIP/testgsm-081784b0 -- Attempting native bridge of SIP/testgsm-081784b0 and SIP/testg729-0817dd90 - After the call is established the UAC is sending some RTP captured in a pcap file in gsm: -- tcpdump -T rtp udp --- 15:58:31.868404 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.868676 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 15:58:31.895551 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.895775 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 15:58:31.936468 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.936477 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.936711 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 15:58:31.936908 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 - Is there something wrong within the SDP? or Am I doing something wrong? Any comments would be appreciated.. thanks!! P.S. I am using Asterisk 1.2.12.1 if that matters. -- Greetings... Víctor Toofic --- 2006-11-15 16:15:12 UDP message sent: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-0 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:34836 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 138 v=0 o=user1 53655765 2353687637 IN IP4 172.16.51.244 s=- c=IN IP4 172.16.51.244 t=0 0 m=audio 10001 RTP/AVP 0 a=rtpmap:18 GSM/8000 --- 2006-11-15 16:15:12 UDP message received [404] bytes : SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-0;received= 172.16.51.244 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- 2006-11-15 16:15:12 UDP message received [609] bytes : SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-0;received= 172.16.51.244 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060;tag=as14685910 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 161 v=0 o=root 3567 3567 IN IP4 172.16.51.215 s=session c=IN IP4 172.16.51.215 t=0 0 m=audio 17050 RTP/AVP 18 a=rtpmap:18 GSM/8000 a=silenceSupp:off - - - - --- 2006-11-15 16:15:12 UDP message sent: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-5 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060;tag=as14685910 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Contact: sip:[EMAIL PROTECTED]:34836 Max-Forwards: 70 Subject: Performance Test
Re: [asterisk-users] Attempting native bridge of
El jue, nov 16 de 2006 a las 18:28 +0530, Vicky comentaba: g729 is not a free codec . YOu have to buy it from digium at rateof $10 per channel license . If you are just using asterisk and havent bought g729 license then asterisk will just do bridging of g729 and wont edit/transcode stream . On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote: I have the following scenario: g729gsm UAS --- * --- UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ¿Am I wrong? I get the same message even if I'm not using g729: --Attempting native bridge of SIP/testgsm-081784b0 and SIP/testulaw-0817da80 ulawgsm UAS --- * --- UAC ___ --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] Attempting native bridge of
El jue, nov 16 de 2006 a las 11:35 -0600, Victor Toofic comentaba: El jue, nov 16 de 2006 a las 18:28 +0530, Vicky comentaba: g729 is not a free codec . YOu have to buy it from digium at rateof $10 per channel license . If you are just using asterisk and havent bought g729 license then asterisk will just do bridging of g729 and wont edit/transcode stream . On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote: I have the following scenario: g729gsm UAS --- * --- UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ¿Am I wrong? I get the same message even if I'm not using g729: --Attempting native bridge of SIP/testgsm-081784b0 and SIP/testulaw-0817da80 ulawgsm UAS --- * --- UAC Ok, sorry for insist. I have registered two ip phones using differents codecs (ulaw g729) and I have audio in both ways, so * is doing transcoding. But I am still getting the log 'Attempting native bridge of'.. so I wonder, What does that really mean? Thanks!! ___ --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] Attempting native bridge of
Thats really strange .. if you have made canreinvite=no then it should not even attampt native bridging and should transcode codecs ..something's fishy here .. Also try to put canreinvite=no in testulaw exntension too . On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote: El jue, nov 16 de 2006 a las 18:28 +0530, Vicky comentaba: g729 is not a free codec . YOu have to buy it from digium at rateof $10 per channel license . If you are just using asterisk and havent bought g729 license then asterisk will just do bridging of g729 and wont edit/transcode stream . On 16/11/06, Victor Toofic [EMAIL PROTECTED] wrote: I have the following scenario: g729gsm UAS --- * --- UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ¿Am I wrong? I get the same message even if I'm not using g729: --Attempting native bridge of SIP/testgsm-081784b0 and SIP/testulaw-0817da80 ulawgsm UAS --- * --- UAC ___ --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] Attempting native bridge of
I have the following scenario: g729gsm UAS --- * --- UAC I am using sipp to generate the calls between the UAC and the UAS and sending some rtp from the UAC, I want * to do transcoding but as I see it is not. As long as I know 'Attempting native bridge' means only passing-through the rtp ¿Am I wrong? The UAC and UAS are registering with * properly: --- sip.conf [testgsm] type=friend host=dynamic username=testgsm context=astertest canreinvite=no disallow=all allow=gsm [testg729] type=friend host=dynamic username=testg729 context=astertest canreinvite=no disallow=all allow=g729 - - dspam*CLI sip show peers Name/username HostDyn Nat ACL Port Status testgsm/testgsm172.16.51.244D 1 Unmonitored testg729/testg729 172.16.51.244D 2 Unmonitored -- Executing Answer(SIP/testgsm-081784b0, ) in new stack -- Executing Wait(SIP/testgsm-081784b0, 1) in new stack -- Executing Dial(SIP/testgsm-081784b0, SIP/testg729) in new stack -- Called testg729 -- SIP/testg729-0817dd90 is ringing -- SIP/testg729-0817dd90 answered SIP/testgsm-081784b0 -- Attempting native bridge of SIP/testgsm-081784b0 and SIP/testg729-0817dd90 - After the call is established the UAC is sending some RTP captured in a pcap file in gsm: -- tcpdump -T rtp udp --- 15:58:31.868404 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.868676 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 15:58:31.895551 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.895775 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 15:58:31.936468 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.936477 IP 172.16.51.244.10001 172.16.51.215.17050: udp/rtp 33 c3 15:58:31.936711 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 15:58:31.936908 IP 172.16.51.215.15424 172.16.51.244.20001: udp/rtp 20 c18 - Is there something wrong within the SDP? or Am I doing something wrong? Any comments would be appreciated.. thanks!! P.S. I am using Asterisk 1.2.12.1 if that matters. -- Greetings... Víctor Toofic --- 2006-11-15 16:15:12 UDP message sent: INVITE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-0 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE Contact: sip:[EMAIL PROTECTED]:34836 Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 138 v=0 o=user1 53655765 2353687637 IN IP4 172.16.51.244 s=- c=IN IP4 172.16.51.244 t=0 0 m=audio 10001 RTP/AVP 0 a=rtpmap:18 GSM/8000 --- 2006-11-15 16:15:12 UDP message received [404] bytes : SIP/2.0 100 Trying Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-0;received=172.16.51.244 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Length: 0 --- 2006-11-15 16:15:12 UDP message received [609] bytes : SIP/2.0 200 OK Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-0;received=172.16.51.244 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060;tag=as14685910 Call-ID: [EMAIL PROTECTED] CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: sip:[EMAIL PROTECTED] Content-Type: application/sdp Content-Length: 161 v=0 o=root 3567 3567 IN IP4 172.16.51.215 s=session c=IN IP4 172.16.51.215 t=0 0 m=audio 17050 RTP/AVP 18 a=rtpmap:18 GSM/8000 a=silenceSupp:off - - - - --- 2006-11-15 16:15:12 UDP message sent: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-5 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL PROTECTED]:5060;tag=as14685910 Call-ID: [EMAIL PROTECTED] CSeq: 1 ACK Contact: sip:[EMAIL PROTECTED]:34836 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 --- 2006-11-15 16:16:15 UDP message sent: BYE sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.51.244:34836;branch=z9hG4bK-1-9 From: testgsm sip:[EMAIL PROTECTED]:5060;tag=1 To: testg729 sip:[EMAIL
[asterisk-users] attempting native bridge on TDM2400
Hello list, I am encountering a bit of a problem in working with incoming calls with a TDM2400 and * 1.2.4; when a call comes in, * will correctly detect the ringing, but will sometimes report multiple Attempting native bridge. What I do is basically that when a call comes in, I dial a different box through the same Zaptel interface and I get logs like this: [call comes in] -- Starting simple switch on 'Zap/10-1' Oct 24 16:55:46 NOTICE[8457]: chan_zap.c:6063 ss_thread: Got event 18 (Ring Begin)... Oct 24 16:55:47 NOTICE[8457]: chan_zap.c:6063 ss_thread: Got event 2 (Ring/Answered)... -- Executing Dial(Zap/10-1, Zap/g1|90|t) in new stack -- Called g1 -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 is ringing -- Zap/1-1 answered Zap/10-1 -- Attempting native bridge of Zap/10-1 and Zap/1-1 -- Attempting native bridge of Zap/10-1 and Zap/1-1 -- Attempting native bridge of Zap/10-1 and Zap/1-1 What I don't get is why * has a need for multiple Attempting native bridge on multiple calls flowing through the same interface. What does this mean? Best regards l. -- Loway Research - Home of QueueMetrics http://queuemetrics.loway.it ___ --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] Attempting native bridge
Hi,
I have problems with two trunks, ZAP3 and
ZAP4. ZAP4 is connected to PSTN line while ZAP3 is connected to analogical
switchboard.
The
system is able to redirect calls from ZAP4 to ZAP3, through an
IVR, but, hanging up doesnt work .
This is the CLI report where you can see, at
the end, the attempt of a native bridge instead of the call hang
up.
-- Playing 'custom/aa_10'
(language 'it')
..
-- Executing
Macro("Zap/4-1", "dialout-trunk|1|45|") in new
stack
-- Executing AGI("Zap/4-1",
"recordingcheck|20060720-151752|1153401458.34") in new
stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/recordingcheck
recordingcheck|20060720-151752|1153401458.34: Outbound recording not
enabled
-- AGI Script recordingcheck
completed, returning 0
-- Executing NoOp("Zap/4-1",
"No recording needed") in new stack
-- Executing
Macro("Zap/4-1", "outbound-callerid|1") in new
stack
-- Executing
DBget("Zap/4-1", "USEROUTCID=AMPUSER//outboundcid") in new
stack
-- DBget:
varname=USEROUTCID, family=AMPUSER,
key=/outboundcid
-- DBget: Value not found in
database.
-- Executing
GotoIf("Zap/4-1", "1?4") in new stack
-- Goto
(macro-outbound-callerid,s,4)
-- Executing
GotoIf("Zap/4-1", "1?6") in new stack
-- Goto
(macro-outbound-callerid,s,6)
-- Executing NoOp("Zap/4-1",
"CallerID set to ") in new stack
-- Executing
SetGroup("Zap/4-1", "OUT_1") in new stack
-- Executing
GotoIf("Zap/4-1", "0?108") in new stack
-- Executing
SetVar("Zap/4-1", "DIAL_NUMBER=45") in new
stack
-- Executing
SetVar("Zap/4-1", "DIAL_TRUNK=1") in new
stack
-- Executing AGI("Zap/4-1",
"fixlocalprefix") in new stack
-- Launched AGI Script
/var/lib/asterisk/agi-bin/fixlocalprefix
-- AGI Script fixlocalprefix
completed, returning 0
-- Executing
SetVar("Zap/4-1", "OUTNUM=45") in new stack
-- Executing Cut("Zap/4-1",
"custom=OUT_1|:|1") in new stack
-- Executing
GotoIf("Zap/4-1", "0?16") in new stack
-- Executing Dial("Zap/4-1",
"ZAP/3/45") in new stack
-- Called
3/45
-- Zap/3-1 answered
Zap/4-1
-- Attempting native bridge
of Zap/4-1 and Zap/3-1
Thanks in
advance.
___
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Re: [Asterisk-Users] Attempting native bridge of
On Apr 12, 2005, at 9:38 PM, snacktime wrote: That would be great if I didn't want * to get out of the media path, but I do. In my case everything works great with the teliax 800 DID, but not with the local number DID. I think it's an issue on their end myself. ___ I didn't want to insinuate that Teliax was in any way sloppy, but they *are* the ITSP I was referring to when I mentioned earlier in this thread that my provider was having issues with native bridging. I raised a ticket with them and they're working on resolving the bug currently, so I think you're in the same boat with me here Mr. Snackie. When I get closure on the ticket I'll send you a note and mention it in this thread for future searchers. /rg ___ 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] Attempting native bridge of
I didn't want to insinuate that Teliax was in any way sloppy, but they *are* the ITSP I was referring to when I mentioned earlier in this thread that my provider was having issues with native bridging. I raised a ticket with them and they're working on resolving the bug currently, so I think you're in the same boat with me here Mr. Snackie. When I get closure on the ticket I'll send you a note and mention it in this thread for future searchers. I emailed them also and got a quick reply. Evidently they were aware of the bridging issue, but were not aware that bridging does work on their toll free DID's while it doesn't on the local number DID's. Hope they get it sorted, as I do like their service and overall quality compared to most other low cost pay as you go providers. Chris ___ 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] Attempting native bridge of
Hello We find an issue when IAX wants to transfer the native bridge. We are using asterisk 1.0.7. Asterisk shows following messages after getting 'answered'. -- Attempting native bridge of IAX2/[EMAIL PROTECTED]/2 and IAX2/xxx.xxx.xxx.xxx:/3 -- Channel 'IAX2/[EMAIL PROTECTED]/2' ready to transfer -- Channel 'IAX2/xxx.xxx.xxx.xxx:/3' ready to transfer -- Releasing IAX2/xxx.xxx.xxx.xxx:xxx/3 and IAX2/[EMAIL PROTECTED]/2 It turns out that both channels hangs up, but the call still continues. However if Attempting native bridge of IAX2/[EMAIL PROTECTED]/2 and IAX2/xxx.xxx.xxx.xxx:/3 never returns, Asterisk doesn't transfer, then everything works fine. Can someone shed lights on this? thank you! steven ___ 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] Attempting native bridge of
On Apr 12, 2005, at 11:21 AM, Xu Wang wrote: Hello We find an issue when IAX wants to transfer the native bridge. We are using asterisk 1.0.7. Asterisk shows following messages after getting 'answered'. -- Attempting native bridge of IAX2/[EMAIL PROTECTED]/2 and IAX2/xxx.xxx.xxx.xxx:/3 -- Channel 'IAX2/[EMAIL PROTECTED]/2' ready to transfer -- Channel 'IAX2/xxx.xxx.xxx.xxx:/3' ready to transfer -- Releasing IAX2/xxx.xxx.xxx.xxx:xxx/3 and IAX2/[EMAIL PROTECTED]/2 It turns out that both channels hangs up, but the call still continues. However if Attempting native bridge of IAX2/[EMAIL PROTECTED]/2 and IAX2/xxx.xxx.xxx.xxx:/3 never returns, Asterisk doesn't transfer, then everything works fine. See: http://lists.digium.com/pipermail/asterisk-users/2005-March/097007.html ...and tell me if this is what you're experiencing. In my case it was a ITSP issue; a bug which is being resolved. You also should ensure you're ANSWERing your channel. I see you mentioned it above but I don't know if you really mean you issued the answer command. /rg ___ 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] Attempting native bridge of
i do have 'answer' to 1st incoming IAX before calling the 2nd IAX. Yes, the log looks almost the same. I have 1800 coming from one vendor, then call through 2nd vendor (it might be the same vendor as the 1st ) to the destination. If 'attempting native brige' is successful, both IAXs are hung up. But the call can still continue. -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Robert Goodyear Sent: Tuesday, April 12, 2005 1:20 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Attempting native bridge of On Apr 12, 2005, at 11:21 AM, Xu Wang wrote: Hello We find an issue when IAX wants to transfer the native bridge. We are using asterisk 1.0.7. Asterisk shows following messages after getting 'answered'. -- Attempting native bridge of IAX2/[EMAIL PROTECTED]/2 and IAX2/xxx.xxx.xxx.xxx:/3 -- Channel 'IAX2/[EMAIL PROTECTED]/2' ready to transfer -- Channel 'IAX2/xxx.xxx.xxx.xxx:/3' ready to transfer -- Releasing IAX2/xxx.xxx.xxx.xxx:xxx/3 and IAX2/[EMAIL PROTECTED]/2 It turns out that both channels hangs up, but the call still continues. However if Attempting native bridge of IAX2/[EMAIL PROTECTED]/2 and IAX2/xxx.xxx.xxx.xxx:/3 never returns, Asterisk doesn't transfer, then everything works fine. See: http://lists.digium.com/pipermail/asterisk-users/2005-March/097007.html ...and tell me if this is what you're experiencing. In my case it was a ITSP issue; a bug which is being resolved. You also should ensure you're ANSWERing your channel. I see you mentioned it above but I don't know if you really mean you issued the answer command. /rg ___ 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 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] Attempting native bridge of
On Apr 12, 2005, at 1:35 PM, Xu Wang wrote: i do have 'answer' to 1st incoming IAX before calling the 2nd IAX. Yes, the log looks almost the same. I have 1800 coming from one vendor, then call through 2nd vendor (it might be the same vendor as the 1st ) to the destination. If 'attempting native brige' is successful, both IAXs are hung up. But the call can still continue. Wait a moment... you mean the call is not dropped, but Asterisk is simply taken out of the media path? If so, I understand this to be the intended behavior unless you invoke the NOTRANSFER=YES directive in your IAX.conf for either channel. This I think is analogous to the CANREINVITE parameter for SIP. I hope someone else will correct me if my understanding is wrong. /rg ___ 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] Attempting native bridge of
It is caused by 'notransfer'. The call is not dropped. thanks! steven -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Robert Goodyear Sent: Tuesday, April 12, 2005 3:00 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Attempting native bridge of On Apr 12, 2005, at 1:35 PM, Xu Wang wrote: i do have 'answer' to 1st incoming IAX before calling the 2nd IAX. Yes, the log looks almost the same. I have 1800 coming from one vendor, then call through 2nd vendor (it might be the same vendor as the 1st ) to the destination. If 'attempting native brige' is successful, both IAXs are hung up. But the call can still continue. Wait a moment... you mean the call is not dropped, but Asterisk is simply taken out of the media path? If so, I understand this to be the intended behavior unless you invoke the NOTRANSFER=YES directive in your IAX.conf for either channel. This I think is analogous to the CANREINVITE parameter for SIP. I hope someone else will correct me if my understanding is wrong. /rg ___ 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 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] Attempting native bridge of
I also have a native bridge problem. I have 2 analogue phones each connected to an IAXy. When attempting a call between them I get the following: -- Accepting DIAL from nnn.nnn.117.75, formats = 0x4 -- Executing Dial(IAX2/[EMAIL PROTECTED]/6, IAX2/kitchen) in new stack -- Called kitchen -- Call accepted by 192.168.6.142 (format ulaw) -- Format for call is ulaw -- IAX2/kitchen/8 is ringing -- IAX2/kitchen/8 answered IAX2/[EMAIL PROTECTED]/6 -- Attempting native bridge of IAX2/[EMAIL PROTECTED]/6 and IAX2/kitchen/8 Apr 12 17:12:18 DEBUG[2053]: chan_iax2.c:5353 socket_read: Ooh, voice format changed to 4 Apr 12 17:12:18 DEBUG[2053]: chan_sip.c:771 __sip_autodestruct: Auto destroying call '[EMAIL PROTECTED]' Apr 12 17:12:18 DEBUG[2053]: db.c:163 ast_db_get: Unable to find key 'si-000fd3000a7e' in family 'iax/provisioning/cache' Apr 12 17:12:18 DEBUG[2053]: iax2-provision.c:231 iax_provision_version: Unable to create provisioning packet for 'si-000fd3000a7e' Apr 12 17:12:19 DEBUG[2053]: chan_iax2.c:5353 socket_read: Ooh, voice format changed to 4 Apr 12 17:12:19 DEBUG[2053]: chan_iax2.c:5835 socket_read: Immediately destroying 8, having received INVAL Apr 12 17:12:19 DEBUG[2053]: channel.c:2630 ast_channel_bridge: Returning from native bridge, channels: IAX2/[EMAIL PROTECTED]/6, IAX2/kitchen/8 Apr 12 17:12:19 DEBUG[2053]: chan_iax2.c:2407 iax2_hangup: We're hanging up IAX2/kitchen/8 now... Apr 12 17:12:19 DEBUG[2053]: chan_iax2.c:2416 iax2_hangup: Really destroying IAX2/kitchen/8 now... -- Hungup 'IAX2/kitchen/8' Apr 12 17:12:19 DEBUG[2053]: app_dial.c:1036 dial_exec: Exiting with DIALSTATUS=ANSWER. == Spawn extension (local, 1133, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/6' -- Executing Hangup(IAX2/[EMAIL PROTECTED]/6, ) in new stack == Spawn extension (local, h, 1) exited non-zero on 'IAX2/[EMAIL PROTECTED]/6' Apr 12 17:12:19 DEBUG[2053]: chan_iax2.c:2407 iax2_hangup: We're hanging up IAX2/[EMAIL PROTECTED]/6 now... -- Hungup 'IAX2/[EMAIL PROTECTED]/6' Apr 12 17:12:27 DEBUG[2053]: chan_iax2.c:5535 socket_read: Immediately destroying 12, having received hangup Apr 12 17:12:27 DEBUG[2053]: chan_iax2.c:3792 raw_hangup: Raw Hangup nnn.nnn.117.75:4569, src=11, dst=520 Apr 12 17:12:27 DEBUG[2053]: chan_iax2.c:3792 raw_hangup: Raw Hangup nnn.nnn.117.75:4569, src=12, dst=7260 Any ideas? Cheers Mike On Wed, 2005-04-13 at 10:00, Robert Goodyear wrote: On Apr 12, 2005, at 1:35 PM, Xu Wang wrote: i do have 'answer' to 1st incoming IAX before calling the 2nd IAX. Yes, the log looks almost the same. I have 1800 coming from one vendor, then call through 2nd vendor (it might be the same vendor as the 1st ) to the destination. If 'attempting native brige' is successful, both IAXs are hung up. But the call can still continue. Wait a moment... you mean the call is not dropped, but Asterisk is simply taken out of the media path? If so, I understand this to be the intended behavior unless you invoke the NOTRANSFER=YES directive in your IAX.conf for either channel. This I think is analogous to the CANREINVITE parameter for SIP. I hope someone else will correct me if my understanding is wrong. /rg ___ 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 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] Attempting native bridge of
On Apr 12, 2005, at 3:31 PM, Mike Price wrote: I also have a native bridge problem. I have 2 analogue phones each connected to an IAXy. When attempting a call between them I get the following: -- Accepting DIAL from nnn.nnn.117.75, formats = 0x4 -- Executing Dial(IAX2/[EMAIL PROTECTED]/6, IAX2/kitchen) in new stack -- Called kitchen -- Call accepted by 192.168.6.142 (format ulaw) -- Format for call is ulaw -- IAX2/kitchen/8 is ringing -- IAX2/kitchen/8 answered IAX2/[EMAIL PROTECTED]/6 -- Attempting native bridge of IAX2/[EMAIL PROTECTED]/6 and IAX2/kitchen/8 snipped Didja try NOTRANSFER=YES ? ___ 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] Attempting native bridge of
I have a very strange bridging problem also with teliax. I have an
800 DID and a local number DID with them. Both numbers go to the same
context, where the caller is dropped into DISA, and the outgoing call
also goes out through teliax.
When dialing into the 800 number, everything works. When dialing into
the local DID, the caller enters the number he wants to call, asterisk
goes through bridging the call, but when asterisk releases the
channels and hangs up the caller hears a second of garbled voice, then
a strange sound like an electic motor dying, and the call is dropped.
extensions.conf:
[outgoing]
exten =
_1XX,1,DIAL(IAX2/snacktime:[EMAIL PROTECTED]/${EXTEN},20,tr)
; I used '_.' just to make sure dialplan for both DID's was in fact the same.
[from-teliax]
exten = _.,1,Ringing
exten = _.,2,Wait(4)
exten = _.,3,Answer
exten = _.,4,Wait(2)
exten = _.,5,Authenticate(1234)
exten = _.,6,Playback(pls-entr-num-uwish2-call)
exten = _.,7,DISA(no-password|outgoing)
iax.conf:
[general]
; Teliax
register = snacktime:[EMAIL PROTECTED]
[teliax]
context=from-teliax
type=friend
host=voip.teliax.com
auth=md5
secret=xx
disallow=all
allow=ulaw
Following is a log off the * console. The log looks the same whether
it's the 800 number or local DID.
Accepting AUTHENTICATED call from 208.139.204.228, requested format =
4, actual format = 4
-- Executing Ringing(IAX2/[EMAIL PROTECTED]/5, ) in new stack
-- Executing Wait(IAX2/[EMAIL PROTECTED]/5, 4) in new stack
-- Executing Answer(IAX2/[EMAIL PROTECTED]/5, ) in new stack
-- Executing Wait(IAX2/[EMAIL PROTECTED]/5, 2) in new stack
-- Executing Authenticate(IAX2/[EMAIL PROTECTED]/5, 1234) in new stack
-- Playing 'agent-pass' (language 'en')
-- Playing 'auth-thankyou' (language 'en')
-- Executing Playback(IAX2/[EMAIL PROTECTED]/5,
pls-entr-num-uwish2-call) in new stack
-- Playing 'pls-entr-num-uwish2-call' (language 'en')
-- Executing DISA(IAX2/[EMAIL PROTECTED]/5, no-password|outgoing)
in new stack
Apr 12 20:52:13 WARNING[971]: cdr.c:286 ast_cdr_init: CDR already
initialized on 'IAX2/[EMAIL PROTECTED]/5'
-- Executing Dial(IAX2/[EMAIL PROTECTED]/5,
IAX2/user:[EMAIL PROTECTED]/12063821515|20|tr) in new stack
-- Called user:[EMAIL PROTECTED]/12063821515
-- Call accepted by 208.139.204.228 (format ulaw)
-- Format for call is ulaw
-- IAX2/teliax/3 is ringing
-- IAX2/teliax/3 answered IAX2/[EMAIL PROTECTED]/5
-- Attempting native bridge of IAX2/[EMAIL PROTECTED]/5 and IAX2/teliax/3
-- Channel 'IAX2/[EMAIL PROTECTED]/5' ready to transfer
-- Channel 'IAX2/teliax/3' ready to transfer
-- Releasing IAX2/teliax/3 and IAX2/[EMAIL PROTECTED]/5
-- Hungup 'IAX2/teliax/3'
== Spawn extension (outgoing, 12063821515, 1) exited non-zero on
'IAX2/[EMAIL PROTECTED]/5'
-- Hungup 'IAX2/[EMAIL PROTECTED]/5'
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RE: [Asterisk-Users] Attempting native bridge of
add following line in the context of IAX.conf
NOTRANSFER=YES
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of snacktime
Sent: Tuesday, April 12, 2005 9:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Attempting native bridge of
I have a very strange bridging problem also with teliax. I have an
800 DID and a local number DID with them. Both numbers go to the same
context, where the caller is dropped into DISA, and the outgoing call
also goes out through teliax.
When dialing into the 800 number, everything works. When dialing into
the local DID, the caller enters the number he wants to call, asterisk
goes through bridging the call, but when asterisk releases the
channels and hangs up the caller hears a second of garbled voice, then
a strange sound like an electic motor dying, and the call is dropped.
extensions.conf:
[outgoing]
exten =
_1XX,1,DIAL(IAX2/snacktime:[EMAIL PROTECTED]/${EXTEN},20,tr)
; I used '_.' just to make sure dialplan for both DID's was in fact the
same.
[from-teliax]
exten = _.,1,Ringing
exten = _.,2,Wait(4)
exten = _.,3,Answer
exten = _.,4,Wait(2)
exten = _.,5,Authenticate(1234)
exten = _.,6,Playback(pls-entr-num-uwish2-call)
exten = _.,7,DISA(no-password|outgoing)
iax.conf:
[general]
; Teliax
register = snacktime:[EMAIL PROTECTED]
[teliax]
context=from-teliax
type=friend
host=voip.teliax.com
auth=md5
secret=xx
disallow=all
allow=ulaw
Following is a log off the * console. The log looks the same whether
it's the 800 number or local DID.
Accepting AUTHENTICATED call from 208.139.204.228, requested format =
4, actual format = 4
-- Executing Ringing(IAX2/[EMAIL PROTECTED]/5, ) in new stack
-- Executing Wait(IAX2/[EMAIL PROTECTED]/5, 4) in new stack
-- Executing Answer(IAX2/[EMAIL PROTECTED]/5, ) in new stack
-- Executing Wait(IAX2/[EMAIL PROTECTED]/5, 2) in new stack
-- Executing Authenticate(IAX2/[EMAIL PROTECTED]/5, 1234) in new stack
-- Playing 'agent-pass' (language 'en')
-- Playing 'auth-thankyou' (language 'en')
-- Executing Playback(IAX2/[EMAIL PROTECTED]/5,
pls-entr-num-uwish2-call) in new stack
-- Playing 'pls-entr-num-uwish2-call' (language 'en')
-- Executing DISA(IAX2/[EMAIL PROTECTED]/5, no-password|outgoing)
in new stack
Apr 12 20:52:13 WARNING[971]: cdr.c:286 ast_cdr_init: CDR already
initialized on 'IAX2/[EMAIL PROTECTED]/5'
-- Executing Dial(IAX2/[EMAIL PROTECTED]/5,
IAX2/user:[EMAIL PROTECTED]/12063821515|20|tr) in new stack
-- Called user:[EMAIL PROTECTED]/12063821515
-- Call accepted by 208.139.204.228 (format ulaw)
-- Format for call is ulaw
-- IAX2/teliax/3 is ringing
-- IAX2/teliax/3 answered IAX2/[EMAIL PROTECTED]/5
-- Attempting native bridge of IAX2/[EMAIL PROTECTED]/5 and IAX2/teliax/3
-- Channel 'IAX2/[EMAIL PROTECTED]/5' ready to transfer
-- Channel 'IAX2/teliax/3' ready to transfer
-- Releasing IAX2/teliax/3 and IAX2/[EMAIL PROTECTED]/5
-- Hungup 'IAX2/teliax/3'
== Spawn extension (outgoing, 12063821515, 1) exited non-zero on
'IAX2/[EMAIL PROTECTED]/5'
-- Hungup 'IAX2/[EMAIL PROTECTED]/5'
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Re: [Asterisk-Users] Attempting native bridge of
On 4/12/05, Xu Wang [EMAIL PROTECTED] wrote: add following line in the context of IAX.conf NOTRANSFER=YES That would be great if I didn't want * to get out of the media path, but I do. In my case everything works great with the teliax 800 DID, but not with the local number DID. I think it's an issue on their end myself. ___ 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] Attempting native bridge
ERROR CONDITION --- -- Executing Dial(SIP/2001-f6c4, SIP/2000|20) in new stack -- Called 2000 -- SIP/2000-0ead is ringing -- SIP/2000-0ead answered SIP/2001-f6c4 -- Attempting native bridge of SIP/2001-f6c4 and SIP/2000-0ead Have searched web and archive w/o good results. Thks in advance for any help, Dave sip.conf [general] port = 5060 bindaddr = 0.0.0.0 allow=all context = bogon-calls externip = nn.nnn.nnn.nnn : Behind router, but External static IP nat=yes [2000] type=friend username=2000 secret=2000 host=dynamic context=from-sip mailbox=2000 [2001] type=friend username=2001 secret=2001 host=dynamic context=from-sip mailbox=2001 ;Also had some of these included, but don't understand ;nat=yes; have in [general] as seems to be req'd ;reinvite=no ;canreinvite=no ;qualify=1000 ;disallow=all ;allow=gsm ;allow=ulaw ;allow=alaw extensions.conf --- [general] static=yes writeprotect=yes [bogon-calls] exten = _.,1,Congestion [from-sip] ; ; Number 2000 - Dave Laptop #1 ; exten = 2000,1,Dial(SIP/2000,20) exten = 2000,2,Voicemail(u2000) exten = 2000,102,Voicemail(b2000) exten = 2000,103,Hangup ; ; Number 2001 - Dave Laptop #2 ; exten = 2001,1,Dial(SIP/2001,20) exten = 2001,2,Voicemail(u2001) exten = 2001,102,Voicemail(b2001) exten = 2001,103,Hangup ___ 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] attempting native bridge error
Hi, Hope somebody has an idea as to what the following means: I am making a call from one xlite client (2000) to another xlite client (2001) via asterisk. The call seems to connect fine and each client comes up as 'connected'. They both have the same codecs enabled and have turned the silence settings to yes. However I can hear any audio. My microphone and speakers are working fine and when i did an ethereal sniff I can see the rtp packets being transmitted. On the asterisk console I am seeing a message attempting native bridge between 2000 and 2001. Has anybody any idea what this could be?? Previous threads have suggested that it is the codecs but toh client support gsm and I have allowed for all the codecs in my sip.conf config file. Thanks, Aisling ---Legal Disclaimer--- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. ___ 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] Attempting native bridge .......
The audio is carried on two RTP streams: one for each direction. Is it possible those streams are being blocked by a firewall or something of the sort? The attempting native bridge message means that Asterisk is bridging the two calls together without doing any codec translation... uLaw to uLaw, for instance. If the two phones were using reinvite you wouldn't see this message because there would be nothing for Asterisk to bridge: the two phones are chatting to one another. On Oct 15, 2004, at 3:34 PM, Brian Weaver wrote: I have two fo the Sipura-2000 boxes, one at a friends house, one here. It used to be working but now we are not getting any audio when the call is picked up. I'm seeing this message when he answer the phone. -- Attempting native bridge of SIP/2204-2b1b and SIP/2203-783a As far as I can tell, it shouldn't be doing this because I have canreinvite=no in the sip.conf for these extensions since we are behind NAT firewalls and the two Sipura boxes cannot talk directly to each other. What am I missing? ___ 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
