Alex Malinovich
2004-Jun-22 14:58 UTC
[Asterisk-Users] Two SIP servers communicating without IAX
I'm working on getting two SIP servers to talk. An Asterisk box and a Zultys MX250 system. So far, things have been working pretty well. I can call from one of my Asterisk-managed phones to an MX250-managed phone and vice versa. However, there are some strange issues. If I call from an MX250 phone to an Asterisk phone, the conversation is ok, but there is a noticeable delay in the voice stream. If I call from an Asterisk phone to an MX250 phone, I can talk FROM the MX250 phone TO the Asterisk phone, but not the other way around. In other words, the Asterisk phone will hear everything that is said from the MX250 phone, but if I say anything on the Asterisk phone the MX250 phone never gets it. Any suggestions? -- Alex Malinovich Golden Technologies, Inc. (219) 462-7200 x 216 http://www.golden-tech.com -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040622/b4e6bc94/attachment.pgp
Philipp von Klitzing
2004-Jun-23 03:01 UTC
[Asterisk-Users] Two SIP servers communicating without IAX
Hi!> If I call from an MX250 phone to an Asterisk phone, the conversation is > ok, but there is a noticeable delay in the voice stream.You will probably want to start by analysing your ethernet network (using ping, traceroute, ethereal etc). You might also want to eliminate a switch in between and directly link the two systems to see if things improve.> If I call from an Asterisk phone to an MX250 phone, I can talk FROM the > MX250 phone TO the Asterisk phone, but not the other way around. In > other words, the Asterisk phone will hear everything that is said from > the MX250 phone, but if I say anything on the Asterisk phone the MX250 > phone never gets it.Look at your codec configuration (disallow=all followed byallow= statements). Do a "SIP DEBUG" on the Asterisk CLI to learn more about the codec negotiation before/during a call. Finally check your phones VAD/ silence suppression settings, make sure those are turned off. Cheers, Philipp
Alex Malinovich
2004-Jun-23 07:26 UTC
[Asterisk-Users] Two SIP servers communicating without IAX
On Wed, 2004-06-23 at 05:01, Philipp von Klitzing wrote: --snip--> > If I call from an Asterisk phone to an MX250 phone, I can talk FROM the > > MX250 phone TO the Asterisk phone, but not the other way around. In > > other words, the Asterisk phone will hear everything that is said from > > the MX250 phone, but if I say anything on the Asterisk phone the MX250 > > phone never gets it. > > Look at your codec configuration (disallow=all followed byallow= > statements). Do a "SIP DEBUG" on the Asterisk CLI to learn more about the > codec negotiation before/during a call. Finally check your phones VAD/ > silence suppression settings, make sure those are turned off.That took care of it. I added a disallow=all followed by an allow=ulaw and it worked fine. What's strange is that I had manually set the codec on the phone earlier to use ulaw, but it didn't appear to want to listen. Forcing it via sip.conf took care of the problem. Thanks a lot for the help. -- Alex Malinovich Golden Technologies, Inc. (219) 462-7200 x 216 http://www.golden-tech.com -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20040623/2ac17c40/attachment.pgp